Release Notes Version 6.0 Document #: LTRT-65615 February 2010 SIP Release Notes Contents Table of Contents 1 What's New in Release 6.0 ................................................................................. 9 1.1 Supported Hardware Platforms .............................................................................. 9 1.1.1 1.1.2 1.1.3 New Models and Hardware Configurations Introduced in this Release ................... 9 Existing Hardware Platforms ..................................................................................... 9 Hardware Platforms No Longer Supported ............................................................... 9 1.2 SIP New Features................................................................................................ 10 1.3 Networking New Features .................................................................................... 21 1.4 Security New Features......................................................................................... 22 1.5 Web New Features .............................................................................................. 22 1.6 SNMP New Features ........................................................................................... 24 1.7 1.8 Miscellaneous New Features ............................................................................... 25 New Parameters .................................................................................................. 26 1.8.1 1.8.2 1.8.3 1.8.4 1.8.5 1.9 SIP Parameters ....................................................................................................... 26 Voice, RTP and RTCP Parameters ........................................................................ 34 Networking Parameters .......................................................................................... 35 Security Parameters................................................................................................ 36 Existing ini File Parameters Now Configurable in the Web .................................... 39 Modified Parameters ............................................................................................ 40 1.9.1 1.9.2 SIP Parameters ....................................................................................................... 40 Voice, RTP and RTCP Parameters ........................................................................ 45 1.10 Obsolete Parameters ........................................................................................... 46 2 Supported Features.......................................................................................... 47 2.1 SIP Features........................................................................................................ 47 2.1.1 2.1.2 2.1.3 2.2 3 Supported SIP Features ......................................................................................... 47 Unsupported SIP Features ..................................................................................... 50 SIP Compliance Tables .......................................................................................... 50 2.1.3.1 SIP Functions .......................................................................................... 50 2.1.3.2 SIP Methods ............................................................................................ 51 2.1.3.3 SIP Headers ............................................................................................ 51 2.1.3.4 SDP Headers ........................................................................................... 53 2.1.3.5 SIP Responses ........................................................................................ 54 DSP Firmware Templates .................................................................................... 58 Known Constraints........................................................................................... 59 3.1 Voice, RTP and RTCP Constraints ...................................................................... 59 3.2 Infrastructure Constraints ..................................................................................... 59 3.3 Networking Constraints ........................................................................................ 60 3.4 3.5 Security Constraints ............................................................................................. 60 Web Constraints .................................................................................................. 61 3.6 SNMP Constraints ............................................................................................... 61 3.7 CLI Constraints .................................................................................................... 61 Version 6.0 3 February 2010 MP-11x & MP-124 4 5 Resolved Constraints ....................................................................................... 63 4.1 Voice, RTP and RTCP Resolved Constraints ...................................................... 63 4.2 Web Resolved Constraints .................................................................................. 63 4.3 SNMP Resolved Constraints................................................................................ 63 Earlier Releases ................................................................................................ 65 SIP Release Notes 4 Document #: LTRT-65615 SIP Release Notes Contents List of Tables Table 1-1: New SIP Parameters for Release 6.0 .................................................................................. 26 Table 1-2: New Voice, RTP and RTCP Parameters for Release 6.0 .................................................... 34 Table 1-3: New Networking Parameters for Release 6.0 ...................................................................... 35 Table 1-4: New Security Parameters for Release 6.0 ........................................................................... 36 Table 1-5: ini File Parameters now Configurable in the Web Interface for Release 6.0 ....................... 39 Table 1-6: Modified SIP Parameters for Release 6.0 ............................................................................ 40 Table 1-7: Modified Voice, RTP and RTCP Parameter for Release 6.0 ............................................... 45 Table 1-8: Obsolete Parameters ............................................................................................................ 46 Table 2-1: Supported SIP Functions...................................................................................................... 50 Table 2-2: Supported SIP Methods ....................................................................................................... 51 Table 2-3: Supported SIP Headers........................................................................................................ 51 Table 2-4: Supported SDP Headers ...................................................................................................... 53 Table 2-5: Supported 1xx SIP Responses ............................................................................................ 54 Table 2-6: Supported 2xx SIP Responses ............................................................................................ 54 Table 2-7: Supported 3xx SIP Responses ............................................................................................ 55 Table 2-8: Supported 4xx SIP Responses ............................................................................................ 55 Table 2-9: Supported 5xx SIP Responses ............................................................................................ 57 Table 2-10: Supported 6xx SIP Responses........................................................................................... 57 Table 2-11: DSP Firmware Template for MediaPack Series ................................................................. 58 Version 6.0 5 February 2010 MP-11x & MP-124 Reader’s Notes SIP Release Notes 6 Document #: LTRT-65615 SIP Release Notes Notices Notice This document describes the release of the AudioCodes MP-11x and MP-124 MediaPack Series of Voice-over-IP (VoIP) media gateways. Information contained in this document is believed to be accurate and reliable at the time of printing. However, due to ongoing product improvements and revisions, AudioCodes cannot guarantee accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions. Updates to this document and other documents can be viewed by registered customers at http://www.audiocodes.com/downloads. © Copyright 2010 AudioCodes Ltd. All rights reserved. This document is subject to change without notice. Date Published: February-02-2010 Trademarks AudioCodes, AC, AudioCoded, Ardito, CTI2, CTI², CTI Squared, HD VoIP, HD VoIP Sounds Better, InTouch, IPmedia, Mediant, MediaPack, NetCoder, Netrake, Nuera, Open Solutions Network, OSN, Stretto, TrunkPack, VMAS, VoicePacketizer, VoIPerfect, VoIPerfectHD, What’s Inside Matters, Your Gateway To VoIP and 3GX are trademarks or registered trademarks of AudioCodes Limited. All other products or trademarks are property of their respective owners. Product specifications are subject to change without notice. WEEE EU Directive Pursuant to the WEEE EU Directive, electronic and electrical waste must not be disposed of with unsorted waste. Please contact your local recycling authority for disposal of this product. Customer Support Customer technical support and service are provided by AudioCodes’ Distributors, Partners, and Resellers from whom the product was purchased. For Customer support for products purchased directly from AudioCodes, contact [email protected]. Abbreviations and Terminology Each abbreviation, unless widely used, is spelled out in full when first used. Only industrystandard terms are used throughout this manual. Hexadecimal notation is indicated by 0x preceding the number. Version 6.0 7 February 2010 MP-11x & MP-124 Related Documentation Manual Name Product Reference Manual for SIP CPE Devices MP-11x & MP-124 SIP Installation Manual MP-11x & MP-124 SIP User's Manual MP-11x SIP Fast Track Guide MP-124 AC SIP Fast Track Guide MP-124 DC SIP Fast Track Guide CPE SIP Configuration Guide for IP Voice Mail Notes: Throughout this manual, unless otherwise specified, the following terms are used: SIP Release Notes • MediaPack or device refers to the MP-124, MP-118, MP-114, and MP112 VoIP gateways. • MP-11x refers to the MP-118, MP-114, and MP-112 MediaPack series VoIP gateways. 8 Document #: LTRT-65615 SIP Release Notes 1 1. What's New in Release 6.0 What's New in Release 6.0 Note: This document uses a one-row table with check boxes convention to indicate the products/interfaces for which each feature is applicable. Only the products/interfaces with marked check boxes are applicable to the feature. For example, the table below indicates that the feature is applicable only to MP-11x FXS. MP-124 MP-11x FXS FXO 1.1 Supported Hardware Platforms 1.1.1 New Models and Hardware Configurations Introduced in this Release Not applicable. 1.1.2 Existing Hardware Platforms The following existing hardware platforms are supported in this release: 1.1.3 MP-11x combined FXS/FXO devices: • MP-114/FXS+FXO providing 2 FXS ports and 2 FXO ports • MP-118/FXS+FXO providing 4 FXS ports and 4 FXO ports MP-11x/FXO devices: • MP-118/FXO providing 8 analog FXO interfaces • MP-114/FXO providing 4 analog FXO interfaces MP-11x/FXS devices: • MP-118/FXS providing 8 analog FXS interfaces • MP-114/FXS providing 4 analog FXS interfaces • MP-112/FXS providing 2 analog FXS interfaces MP-124/FXS providing 24 analog FXS interfaces MP-124 with AC Power MP-124 with DC power Hardware Platforms No Longer Supported Not applicable. Version 6.0 9 February 2010 MP-11x & MP-124 1.2 SIP New Features The device supports the following new SIP features: 1. Sending SIP Response to UDP Port from where SIP Request Received regardless of "rport" Parameter: MP-124 MP-11x FXS FXO In previous releases, the device sent SIP responses to the UDP port defined in the SIP Via header. If the Via header contained the "rport" parameter, the device sent the response to the UDP port from where the SIP request was received. In this release, the device can be configured (using the new parameter, SIPForceRport) to send SIP responses to the UDP port from where the SIP request was received even if the "rport" parameter is not received in the Via header. Relevant parameter: SIPForceRport. 2. BYE after SIP 202 Accepted for Blind Transfer: MP-124 MP-11x FXS FXO In previous releases, after initiating a Blind Transfer, the device's FXS endpoint was busy until receipt of a SIP NOTIFY with 200 OK. The receipt of this NOTIFY could take a few minutes after sending a REFER message. During this period, the device could make a new call from the same port, but could not perform a new Blind Transfer. In this release, when initiating a Blind Transfer (using the DTMF KeyBlindTransfer code), the device now sends a BYE message upon receipt of a SIP 202 Accepted response, thereby terminating the REFER dialog session. This allows the FXS endpoint to make a new Blind Transfer without having to wait for a NOTIFY with 200 OK response. 3. Offered Coders Increased to 10: MP-124 MP-11x FXS FXO The device now supports the configuration of up to 10 coders (compared to only 5 in the previous release) for offering the remote end. This also applies to Coder Groups, where up to 10 coders can now be defined per Coder Group (compared to only 5 in the previous release). In addition, a new parameter, CodersGroup now replaces the CoderName parameter (from previous versions). This new parameter supports backward compatibility, allowing users from previous versions to seamlessly upgrade to Version 6.0 (the coders defined under the CoderName parameter are transferred to the CodersGroup parameter). Relevant parameters: CodersGroup; CoderName SIP Release Notes 10 Document #: LTRT-65615 SIP Release Notes 4. 1. What's New in Release 6.0 Tel-to-IP Redirect Number Manipulation: MP-124 MP-11x FXS FXO The device now supports Tel-to-IP Redirect Number manipulation, configured using the new table, Redirect Number Tel to IP table. This feature allows you to manipulate the prefix of the redirect number received from the PSTN for the outgoing SIP Diversion, Resource-Priority, or History-Info header that is sent to IP. Relevant parameter: RedirectNumberMapTel2Ip. 5. IP-to-Tel Call Forwarding to IP Destination upon Unavailable Hunt Group: MP-124 MP-11x FXS FXO The device now supports the forwarding of IP-to-Tel calls to a different IP destination, using SIP 3xx response if an unavailable FXS/FXO Hunt Group exists. This feature can be used, for example, to forward the call to another FXS/FXO device. This feature is configured using the new table, Forward On Busy Trunk Destination, which defines an alternative IP destination (IP address, port and transport type) per Hunt Group. The device forwards calls using this new table only if no alternative IP-to-Tel routing has been configured or alternative routing fails, and the following reason (included in the SIP Diversion header of 3xx messages) exists: • "unavailable": all FXS/FXO lines pertaining to a Hunt Group are busy or unavailable Relevant parameter: ForwardOnBusyTrunkDest. 6. FXS Distinctive Ringing and Call Waiting Tones per Calling Number for IP-to-Tel Calls: MP-124 MP-11x FXS FXO The device now supports the configuration of a Distinctive Ringing tone and Call Waiting Tone per calling number for IP-to-Tel calls. This feature can be configured per FXS endpoint or for a range of FXS endpoints. Therefore, different tones can be played per FXS endpoint/s, depending on the source number of the received call. This configuration is performed in a new table that maps Ringing and/or Call Waiting tones to source number prefixes, per FXS endpoint/s. Typically, the Ringing and/or Call Waiting tone played is indicated in the SIP Alert-info header field of the received INVITE message. If this header is not present in the received INVITE, then this feature is used and the tone played is according to the settings in this new table. Relevant parameter: ToneIndex. 7. Version 6.0 11 February 2010 MP-11x & MP-124 Interworking SAS behind NAT: MP-124 MP-11x FXS FXO Since SAS is implemented as a standard proxy, it’s default behavior while relaying requests is as follows: • Adds a new SIP Via header (with the SAS IP address) as the top-most Via header. • Does not modify the original SIP Contact header. This default and standard proxy result in the top-most Via header and the Contact header to point to different hosts. However, some SBC’s (e.g., ACME) require that incoming requests must point to the same host in the top-most Via header and the Contact header. For interoperability support with such an SBC, the device can now operate in a new mode that changes the Contact header so that it points to the SAS host, and therefore, the top-most Via header and the Contact header point to the same host. Note that operating in this mode causes all incoming dialog requests to traverse the SAS, and thus, may cause load problems. Parameter: SASEnableContactReplace. 8. SIP Record-Route Header for SAS: MP-124 MP-11x FXS FXO The device's SAS application now can be configured to add the SIP Record-Route header to SIP requests. This feature ensures that SIP messages traverse the device's SAS agent, by including the SAS IP address in the Record-Route header. The Record-Route header is inserted in a request by a proxy server to force future requests in the dialog session to be routed through the proxy. Each traversed proxy in the path can insert this header, causing all future dialogs in the session to pass through it as well. When this feature is enabled, the SIP Record-Route header includes the URI "lr" parameter. The presence of this parameter indicates loose routing; the lack of 'lt' indicates strict routing. For example: • Loose routing: Record-Route: <sip:server10.biloxi.com;lr> • Strict routing: Record-Route: <sip:bigbox3.site3.atlanta.com> Relevant parameter: SASEnableRecordRoute. 9. SIP Release Notes 12 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 SAS IP2IP Routing Table for SAS Normal Mode: MP-124 MP-11x FXS FXO The device's SAS application now uses the SAS IP2IP Routing table for routing requests received from the active proxy when in SAS Normal mode (previously used only for SAS Emergency mode). When SAS receives a SIP INVITE request from an active proxy server, the following routing logic is performed: a. Sends the request according to rules configured in the IP2IP Routing table. b. If no matching routing rule exists, the device sends the request according to its SAS registration database. c. If no routing rule is located in the database, the device sends the request according to the Request-URI header. 10. SAS in Normal Mode Responds to REGISTER Requests with 200 OK without Relaying them to Proxy: MP-124 MP-11x FXS FXO The device's SAS application when in Normal mode can now be configured to respond to REGISTER requests by sending a SIP 200 OK (instead of relaying the registration requests to a proxy) and entering the registrations in the SAS database. This new feature is enabled by the new option "Auto-answer REGISTER" (3) for the existing SASSurvivabilityMode parameter. Relevant parameter: SASSurvivabilityMode. 11. Parameters Added to Tel Profile: MP-124 MP-11x FXS FXO The following parameters have now been added to the device's Tel Profile feature: • Enable911PSAP - representing the global parameter Enable911PSAP • SwapTelToIpPhoneNumbers - representing the global parameter SwapTEl2IPCalled&CallingNumbers • EnableAGC - representing the global parameter EnableAGC • ECNlpMode - representing the global parameter ECNlpMode Relevant parameter: TelProfile. 12. DSCP Based on Resource-Priority Header upon Receipt of SIP UPDATE: MP-124 MP-11x FXS FXO Upon receipt of the SIP UPDATE (with or without SDP), the device populates the Differentiated Service Code Point (DSCP) markings in the session media stream packets, based on the received precedence level from the SIP Resource-Priority header. Version 6.0 13 February 2010 MP-11x & MP-124 13. Hiding SIP Passwords: MP-124 MP-11x FXS FXO The device now hides configured SIP passwords. Passwords are configured in the 'Proxy & Registration', 'Account Table', and 'Authentication' (for endpoint authentication) pages. Once you configure a password in the Web interface (and the Submit button is clicked), the Web GUI displays the entered password as an asterisk (*). If you load an ini file that includes a configured password, the Web GUI also displays it as an asterisk. When you save an ini file to a PC, the global parameter Password and its value are not displayed in the file. If a password is defined in a table ('Account Table' or 'Authentication'), the saved ini file displays the password value as an asterisk. Note: If the Password parameter has an asterisk as its value and is loaded to the device, it has no affect on the device's configuration (i.e., the existing value of the Password parameter is retained). 14. Call Forward Reminder Tone in Off Hook: MP-124 MP-11x FXS FXO The device now supports playing a special dial tone (Stutter Dial tone – Tone Type 15) to a specific FXS endpoint when the phone is off-hooked and when a third-party Application server (AS), e.g., a softswitch is used to forward calls, intended for the endpoint, to another destination. This is useful in that it reminds the FXS user of this service. This feature does not involve device subscription (SIP SUBSCRIBE) to the AS, and activation/deactivation of the service is notified by the server. An unsolicited NOTIFY request is sent from the AS to the device when the Call Forward service is activated or cancelled. Depending on this NOTIFY request, the device plays the standard dial tone or the special dial tone for Call Forward. For playing the special dial tone, the received SIP NOTIFY message must include the following headers: • From and To headers contain the same information, which indicates the specific endpoint • Event: ua-profile • Content-Type: "application/simservs+xml" • Message body is the XML body and contains the “dial-tone-pattern” set to "special-condition-tone" (<ss:dial-tone-pattern>special-condition-tone</ss:dialtone-pattern>), which is the special tone indication For cancelling the special dial tone and playing the regular dial tone, the received SIP NOTIFY message must include the following headers: • From and To headers contain the same information, which indicates the specific endpoint • Event: ua-profile • Content-Type: "application/simservs+xml" • Message body is the XML body containing the “dial-tone-pattern” set to "standardcondition-tone" (<ss:dial-tone-pattern>standard-condition-tone</ss:dial-tonepattern>), which is the regular dial tone indication SIP Release Notes 14 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 Therefore, the special dial tone is valid until another SIP NOTIFY is received that instructs otherwise (as described above). Note that if the MWI service is active, the MWI dial tone overrides this special Call Forward dial tone. 15. Enhanced Call Forking Support: MP-124 MP-11x FXS FXO The device now allows the configuration of a timeout (in seconds) that is started once the first SIP 2xx response has been received for a User Agent when a proxy server performs call forking (proxy server forwards the INVITE to multiple SIP User Agents). The device sends a SIP ACK and BYE in response to any additional SIP 2xx received from the proxy within this timeout. Once this timeout elapses, the device ignores subsequent SIP 2xx responses. In addition, the number of supported forking calls per channel has been increased from 4 to 20. In other words, the device can now receive up to 20 forking responses from a single INVITE message. Relevant parameter: ForkingTimeOut. 16. Routing IP-to-Tel Calls to Specific Hunt Groups According to CIC Parameter in Request URI: MP-124 MP-11x FXS FXO The device now supports IP-to-Tel routing decisions based on the SIP carrier identification code ("cic") parameter. It uses the "cic" parameter in the incoming SIP INVITE message to route the call to a specific Hunt Group. For supporting this new feature, this release introduces the new parameter, AddCicAsPrefix. When this parameter is enabled, the device adds the "cic" prefix to the destination number (for IP-to-Tel calls). For example: INVITE sip:5550001;[email protected]:5060;user=phone SIP/2.0 After number manipulation performed by the device, the destination number results in "cic+167895550001". Note: After the cic prefix is added, the Inbound IP Routing table can be used to route this call to a specific Trunk Group. The Destination Number IP to Tel Manipulation table must be used to remove this prefix before placing the call to the Tel. Relevant parameter: AddCicAsPrefix. 17. Version 6.0 15 February 2010 MP-11x & MP-124 SIP "dtg" Parameter for Routing IP-to-Tel Calls to Hunt Groups: MP-124 MP-11x FXS FXO The device now supports the "dtg" parameter for defining the Hunt Group for routing IP-to-Tel calls. This parameter can be used instead of the "tgrp/trunk-context" parameters. The "dtg" parameter appears in the INVITE, for example: INVITE sip:[email protected];dtg=56;user=phone SIP/2.0 The "dtg" parameter also appears in the SIP To header. This feature is enabled by the new parameter, UseBroadsoftDTG (set to 1). If the Hunt Group is not found based on the "dtg" parameter, the IP to Trunk Group Routing table is used instead for routing the call to the appropriate Trunk Group. Relevant parameter: UseBroadsoftDTG. 18. IP-to-Tel Routing Precedence using "tgrp"/"dtg" Parameters or IP to Hunt Group Table: MP-124 MP-11x FXS FXO In previous releases, IP-to-Tel routing was determined by the IP to Hunt Group Routing table (PSTNPrefix ini file parameter), and only if a matching rule was not found in this table did the device use the "tgrp"/"dtg" parameters for routing the call. However, in this release, you can change this priority so that the device first places precedence on the tgrp/dtg parameters for IP-to-Tel routing. If the received INVITE request URI does not contain the tgrp/dtg parameters, or if the Hunt Group number is not defined, then the IP to Hunt Group Routing table is used for routing the call. The IP-to-Tel Routing Precedence feature is enabled using a new parameter, TGRProutingPrecedence. If set to 1, the device performs routing according to the tgrp/dtg parameters. If set to 0 (default), the behavior is the same as in previous releases (first locates a match in the routing table and only if not found, attempts to route the call according to the tgrp parameter). Below is an example of an INVITE request URI with the tgrp parameter, indicating that the IP call should be routed to Hunt Group 7: INVITE sip:200;tgrp=7;[email protected];user=phone SIP/2.0 Note that the UseSIPTgrp parameter must be set to 2 for enabling routing based on the SIP tgrp parameter. Relevant parameters: UseSIPTgrp; TGRProutingPrecedence. 19. SIP Release Notes 16 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 Fake Retry-After Header: MP-124 MP-11x FXS FXO This feature enables the device to operate with proxy servers that do not include the Retry-After SIP header in SIP 503 (Service Unavailable) responses to indicate an unavailable service. The Retry-After header is used with the 503 (Service Unavailable) response to indicate how long the service is expected to be unavailable to the requesting SIP client. The value of this field can either be an HTTP-date or an integer number of seconds (in decimal) after the time of the response. The device maintains a list of available proxies, by using the Keep-Alive (KA) mechanism. The device checks the availability of proxies by sending SIP OPTIONS every KA timeout to all potential proxies. However, some third-party media servers reply to SIP OPTIONS even if they are unavailable. In such cases, the third-party server rejects the SIP INVITE, by sending a 503 (Service Unavailable) response. As a result, the device performs a failover and must periodically retry the availability of the server, by sending new calls to it to detect the possibility that the anomaly condition has been cleared. In previous releases, upon receipt of a SIP 503 response, the device discarded the call and the proxy remained in the “live” proxy list, since it responded to the device's SIP OPTIONS. Unless the 503 response included a Retry-After response-header, the device did not send new call to the proxy for a period specified in the header. Therefore, for third-party media servers that do not support the Retry-After responseheader, this release introduces a new parameter, FakeRetryAfter to resolve this issue. If this parameter is set to a positive value (in seconds), when the device receives a 503 response without a Retry-After response-header, it behaves as if the 503 response included a Retry-After response-header with the period specified by this parameter. If this parameter is set to zero, this "Fake Retry-After" feature is disabled. Relevant parameter: FakeRetryAfter. 20. Increased Number of SIP URIs in Received 302 Contact Header: MP-124 MP-11x FXS FXO The device now supports the receipt of up to eight SIP Uniform Resource Identifiers (URIs) in the received 302 Contact header. This feature allows the device to handle the received redirection (302) response messages from the proxy with one or more contacts in one or more Contact headers (for example, Contact: [email protected], [email protected]). The device uses the URIs in the Contact header, which could be one per Contact header or one Contact header could have multiple URIs, to formulate one or more new outbound call requests. 21. Version 6.0 17 February 2010 MP-11x & MP-124 SRTP Option without SDP Capability Negotiation: MP-124 MP-11x FXS FXO In previous releases, the device supported two security modes (configured by the parameter MediaSecurityBehaviour): • Mandatory mode: The device initiates encrypted calls, but if negotiation of the cipher suite fails, the call is terminated. Incoming calls that do not include encryption information are rejected. • Preferable mode: The device initiates encrypted calls. If negotiation of the cipher suite fails, an un-encrypted call is established. Incoming calls that do not include encryption information are accepted (default). In this mode, the device initiates SDP with two media lines (m=) - one for RTP and one for SRTP. In this release, the device supports an additional mode, "Preferable - Single Media", also configured by the existing MediaSecurityBehaviour parameter. This mode is the same as the Preferable mode, except for the following differences: • Instead of two "m=" lines in the suggested SDP, it uses only a single “m=“ line. • Instead of a media line with RTP/SAVP, it uses RTP/AVP. In addition, in this mode, if the remote SIP UA does not support SRTP, it ignores the crypto lines. An example of an SDP with one "m=" line and crypto, v=0 o=AudiocodesGW 1772695605 1772695471 IN IP4 10.33.4.126 s=Phone-Call c=IN IP4 10.33.4.126 t=0 0 m=audio 6000 RTP/AVP 4 0 70 96 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:70 EG711A/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:30 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9LxQeM1/DGtlN2EHp46jfUXrPgpxdWpU/BmSVu9L|2^31 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:5ORhfgoJN8OnjoORISZNRCpegbhMV5D3Ji9wQbp4|2^31 This feature can also be assigned to an IP Profile. Note that for this feature to be functional, the EnableMediaSecurity parameter must be set to 1. Relevant parameters: MediaSecurityBehaviour; IPProfile. SIP Release Notes 18 Document #: LTRT-65615 SIP Release Notes 22. 1. What's New in Release 6.0 Mapping Additional SIT Tones to Q.850 Causes: MP-124 MP-11x FXS FXO Until now, the device was capable of detecting and reporting the following Special Information Tones (SIT) types from the PSTN: • SIT-NC (No Circuit found) • SIT-IC (Operator Intercept) • SIT-VC (Vacant Circuit - non-registered number) • SIT-RO (Reorder - System Busy) These four SIT tones were mapped to Q.850 cause, using the SITQ850Cause parameter (set to 34, by default). In this release, the device now also supports the detection of an additional three SIT tones (which are detected as one of the above SIT tones): • The NC* SIT tone - detected as NC • The RO* SIT tone - detected as RO • The IO* SIT tone - detected as VC The device can now map each of these SIT tones to a Q.850 cause and then map them to SIP 5xx/4xx responses, using the parameters SITQ850CauseForNC, SITQ850CauseForIC, SITQ850CauseForVC, and SITQ850CauseForRO. Note that if these parameters are not used (default), the SIT specific tone is mapped according to the configuration of the SITQ850Cause parameter. The SIT tones and their frequency durations reported by the device are shown in the table below: Special Information Tones (SITs) Name Description First Tone Frequency Duration Second Tone Frequency Duration Third Tone Frequency Duration (Hz) (ms) (Hz) (ms) (Hz) (ms) NC No circuit found 985.2 380 1428.5 380 1776.7 380 IC Operator intercept 913.8 274 1370.6 274 1776.7 380 VC Vacant circuit (non registered number) 985.2 380 1370.6 274 1776.7 380 RO Reorder (system busy) 913.8 274 1428.5 380 1776.7 380 NC* - 913.8 380 1370.6 380 1776.7 380 RO* - 985.2 274 1370.6 380 1776.7 380 IO* - 913.8 380 1428.5 274 1776.7 380 Relevant parameters: SITQ850CauseForNC; SITQ850CauseForIC; SITQ850CauseForVC; SITQ850CauseForRO; SITQ850Cause. Version 6.0 19 February 2010 MP-11x & MP-124 23. Selecting SIP Header for IP-to-Tel Destination Number: MP-124 MP-11x FXS FXO The device now supports selecting the SIP header for obtaining the called (destination) number (for IP-to-Tel calls). The device can be configured, using the new parameter SelectSourceHeaderForCalledNumber (replacing now obsolete IsUseToHeaderAsCalledNumber parameter), to use one of the following headers for obtaining the destination number: • Request-URI (default) • To • P-Called-Party-ID Relevant parameter: SelectSourceHeaderForCalledNumber. 24. Forced Expiration (SIP Unregistration) using Contact Header Value "*": MP-124 MP-11x FXS FXO The device now supports the removal of SIP UA registration bindings in a Registrar, according to RFC 3261. Registrations are soft state and expire unless refreshed, but can also be explicitly removed. A client can attempt to influence the expiration interval selected by the registrar. A UA requests the immediate removal of a binding by specifying an expiration interval of "0" for that contact address in a REGISTER request. UAs should support this mechanism so that bindings can be removed before their expiration interval has passed. The REGISTER-specific Contact header field value of "*" applies to all registrations, but it can only be used if the Expires header field is present with a value of "0". Use of the "*" Contact header field value allows a registering UA to remove all bindings associated with an address-of-record (AOR) without knowing their precise values. This feature is supported by the introduction of the new parameter, UnRegistrationMode. Relevant parameter: UnregistrationMode. 25. Maximum Proxy Sets Increased to 10: MP-124 MP-11x FXS FXO The device now allows you to configure up to 10 Proxy Sets (compared to only 6 in previous releases). These are configured in the 'Proxy Sets' table (ProxySet). Relevant Parameter: ProxySet. 26. Tel-to-IP Destination Number Manipulation Entries Increased to 120: MP-124 MP-11x FXS FXO The device now allows you to configure up to 120 Tel-to-IP destination number manipulation rules (compared to 100 in the previous release). These are configured in the 'Destination Phone Number Manipulation Table for Tel to IP Calls' table (NumberMapTel2IP). Relevant parameter: NumberMapTel2IP. SIP Release Notes 20 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 27. Sending Re-INVITE with New SRTP Key upon Receipt of 181 Response: MP-124 MP-11x FXS FXO The device now can be configured to send a Re-INVITE with a new SRTP key upon receipt of a SIP 181 response ("call is being forwarded"). This is in accordance with the UCR 2008 standard. If the device sends an INVITE with SDP and receives a 181 response, it changes the SRTP key by sending a Re-INVITE with a new SRTP key in its SDP. Relevant parameter: EnableRekeyAfter181. 1.3 Networking New Features The device supports the following new networking features: 1. TFTP Automatic Provisioning using DHCP Option 55: MP-124 MP-11x FXS FXO The device now supports configuring DHCP Option 55 to include DHCP Options 66 and 67, effectively requesting the DHCP server for TFTP provisioning parameters. You can determine whether to include these options in DHCP Option 55, using a new parameter DHCPRequestTFTPParams. Relevant parameter: DHCPRequestTFTPParams. Version 6.0 21 February 2010 MP-11x & MP-124 1.4 Security New Features The device supports the following new security features: 1. IPSec/IKE Unified Configuration Table: MP-124 MP-11x FXS FXO The device now supports the combined configuration of the Internet Key Exchange (IKE) and IP Security (IPSec) protocols using a single ini file parameter table. This allows for quick and easy configuration (as well as diagnoses) of up to 20 peers. In addition, the user can now use a separate new ini file parameter table for configuring up to four IKE proposal settings, where each proposal defines an encryption algorithm, an authentication algorithm, and a Diffie-Hellman group identifier. Relevant parameters: IPSecSATable; IPSecProposalTable. 2. IEEE 802.1X Port-Based Security: MP-124 MP-11x FXS FXO The device now supports IEEE 802.1X LAN security. The device can function as an IEEE 802.1X supplicant. IEEE 802.1X is a standard for port-level security on secure Ethernet switches; when a device is connected to a secure port, no traffic is allowed until the identity of the device is authenticated. This feature can be configured using the new Web interface page, '802.1x Settings', and SNMP using the new folder acSys802dot1x (OID 1.3.6.1.4.1.5003.9.10.10.1.7.25). The device supports the following Extensible Authentication Protocol (EAP) variants: • MD5-Challenge (EAP-MD5) • Protected EAP (PEAPv0 with EAP-MSCHAPv2) • EAP-TLS Relevant parameters: 802.1xMode; 802.1xUsername; 802.1xPassword; 802.1xVerifyPeerCertificate. 1.5 Web New Features The device supports the following new Web interface features: 1. 'EtherDiscover Operation Mode' Parameter Removed from Web: MP-124 MP-11x FXS FXO The 'EtherDiscover Operation Mode' (EtherDiscoverMode) parameter (appearing in the 'General Security Settings' page) has now been removed from the Web interface. SIP Release Notes 22 Document #: LTRT-65615 SIP Release Notes 2. 1. What's New in Release 6.0 SCE Parameter Removed from Web: MP-124 MP-11x FXS FXO The 'SCE' parameter (appearing in the 'IP Profile Settings' page) has now been removed from the Web interface. 3. Status of Registration per Account: MP-124 MP-11x FXS FXO The Web interface now displays registration status per Account. This information is displayed in a new table in the existing 'Registration Status' page (Status & Diagnostics tab > Gateway Statistics menu > Registration Status). 4. Status of Active Proxy in Proxy Set: MP-124 MP-11x FXS FXO The Web interface now displays the status of the active proxy defined for a Proxy Set. This information is displayed in a new table (Active Proxy Sets Status) in the existing 'Call Routing Status' page (Status & Diagnostics tab > Gateway Statistics menu > Call Routing Status). 5. Software Upgrade Progress Bar Indication: MP-124 MP-11x FXS FXO The device's Web interface now displays a progress bar in the 'Software Upgrade Wizard' page for indicating the progress in real-time of the software file download process. 6. Selectable FXS/FXO Coefficient Files - USA or Europe: MP-124 MP-11x FXS FXO The device's Web interface now supports two new parameters for selecting the required FXS and FXO Coefficient files. The optional file types that the user can select for these parameters are either 'USA' or 'Europe'. This feature replaces the previous option to upload FXO/FXS Coefficient files (in the Web interface - 'Load Auxiliary Files'). To support this new feature, two new parameters have been added to the 'Analog Settings' page (formerly called the 'Hook-Flash Settings' page), under the Media Settings menu. Relevant parameters: FXSCountryCoefficients; CountryCoefficients. Version 6.0 23 February 2010 MP-11x & MP-124 1.6 SNMP New Features The device supports the following new Simple Network Management Protocol (SNMP) features: 1. FXS Coefficient File - USA or European: MP-124 MP-11x FXS FXO The device now supports a new MIB (acAnalogMiscCountyCoefficients) that allows the user to choose between two FXS Coefficient options - USA or Europe. As a result of the introduction of this new MIB, the following existing MIB objects have now been "deprecated": 2. • acAnalogFxoCountryCoefficients. • acSysHTTPClientFXSCoeffFileURL. • acSysHTTPClientFXOCoeffFileURL. Call Pickup: MP-124 MP-11x FXS FXO The device now supports a new MIB parameter miscCallPickup (OID 1.3.6.1.4.1.5003.9.10.3.1.1.11.26) for Call Pickup. 3. Deprecated Parameters: MP-124 MP-11x FXS FXO The following SNMP MIB objects have now been deprecated: • • SIP Release Notes AC-CONTROL-MIB: ♦ acCPNamingEndPoint ♦ acCPNamingTrunk ♦ acCPNamingEndpointPrefix ♦ acMCNamePatternLogicalATM ♦ acMCNameNumberPhysicalEndpointMin ♦ acMCNameNumberStreamEndpointATMStart ♦ acMCProfileBinary AC-SYSTEM-MIB: ♦ acSysHTTPClientFXSCoeffFileURL. ♦ acSysHTTPClientFXOCoeffFileURL. ♦ acSysVLANNetworkServiceClassPriority. ♦ acSysVLANPremiumServiceClassMediaPriority. ♦ acSysVLANGoldServiceClassPriority. ♦ acSysVLANBronzeServiceClassPriority. ♦ acSysVLANPremiumServiceClassControlPriority. 24 Document #: LTRT-65615 SIP Release Notes 4. 1. What's New in Release 6.0 ♦ acSysIKEPolicyTable ♦ acSysIPSecSPDTable Hotline Option Added for sipMiscUseSIPTgrp: MP-124 MP-11x FXS FXO The SNMP MIB parameter sipMiscUseSIPTgrp now provides option value 3, "hotline". The Object Identifier (OID) is 1.3.6.1.4.1.5003.9.10.3.1.2.7.19 and the full path is: iso(1).org(3).dod(6).internet(1).private(4).enterprises(1).audioCodes(5003).acProducts (9).acBoardMibs(10).acGateway(3).gwConfiguration(1).sip(2).sipMisc(7).sipMiscUseSI PTgrp(19). This parameter determines whether the SIP 'tgrp' tag is used, which specifies the Hunt Group to which the call belongs, according to RFC 4904. For example: INVITE sip::+16305550100;tgrp=1;[email protected];user=phone SIP/2.0 5. Future Release 6.2 - Change in Enumeration Names: MP-124 MP-11x FXS FXO In the next major release - Release 6.2 - all hyphens (" - ") will be removed from labels of named-number enumeration. This is in accordance with RFC 2578 SMIv2, which prohibits the use of hyphens. The length of labels will also be shortened to ensure that they are less than 32 characters. 1.7 Miscellaneous New Features The device supports the following miscellaneous new features: 1. Software Upgrade Key: MP-124 MP-11x FXS FXO The device now allows you to upgrade or change the device's supported features, by loading a new (purchased) Software Upgrade Key to match your requirements. The device is supplied with a Software Upgrade Key, which determines the device's supported features, capabilities, and available resources. The Software Upgrade Key is an encrypted key, provided in string format in a text-based file. The Software Upgrade Key can be loaded using the device's Web interface or BootP/TFTP. 2. Automatic Syslog Debug Level Selection Based on CPU Usage: MP-124 MP-11x FXS FXO The device now supports a new debug level of 7. When set to this level, the Syslog level automatically changes between level 5, level 1 and level 0, depending on the device's CPU consumption. In addition, to improve device performance, several Syslog messages are now merged and sent as a single UDP datagram. A new parameter, MaxBundleSyslogLength defines the maximum size of this bundled UDP packet. Relevant parameters: GWDebugLevel; MaxBundleSyslogLength. Version 6.0 25 February 2010 MP-11x & MP-124 1.8 New Parameters This section describes the new parameters for Release 6.0. These new parameters can be configured using the ini file parameter (enclosed in square brackets) and/or the corresponding Web interface parameter (if supported). 1.8.1 SIP Parameters The table below describes the new SIP parameters for Release 6.0. Table 1-1: New SIP Parameters for Release 6.0 Parameter Description Web: Redirect Number Tel -> IP [RedirectNumberMapTel 2IP] This ini file table parameter manipulates the redirect number for Tel-toIP calls. The manipulated Redirect Number is sent in the SIP Diversion, History-Info, or Resource-Priority headers. The format of this parameter is as follows: [RedirectNumberMapTel2Ip] FORMAT RedirectNumberMapTel2Ip_Index = RedirectNumberMapTel2Ip_DestinationPrefix, RedirectNumberMapTel2Ip_RedirectPrefix, RedirectNumberMapTel2Ip_NumberType, RedirectNumberMapTel2Ip_NumberPlan, RedirectNumberMapTel2Ip_RemoveFromLeft, RedirectNumberMapTel2Ip_RemoveFromRight, RedirectNumberMapTel2Ip_LeaveFromRight, RedirectNumberMapTel2Ip_Prefix2Add, RedirectNumberMapTel2Ip_Suffix2Add, RedirectNumberMapTel2Ip_IsPresentationRestricted, RedirectNumberMapTel2Ip_SrcTrunkGroupID, RedirectNumberMapTel2Ip_SrcIPGroupID; [\RedirectNumberMapTel2Ip] For example: RedirectNumberMapTel2Ip 1 = *, 4, 255, 255, 0, 0, 255, , 972, 255, 1, 2; Notes: SIP Release Notes This parameter table can include up to 20 indices (1-20). If the table's matching characteristics rule (i.e., DestinationPrefix, RedirectPrefix, SrcTrunkGroupID, and SrcIPGroupID) is located for the Tel-to-IP call, then the redirect number manipulation rule (defined by the other parameters) is applied to the call. The following parameters are not applicable: NumberType, NumberPlan, and IsPresentationRestricted. The manipulation rules are performed in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and then Suffix2Add. 26 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 Parameter Description Web: Forward On Busy Trunk Destination [ForwardOnBusyTrunkD est] This ini file table parameter configures the Forward On Busy Trunk Destination table. This table allows you to define an alternative IP destination (IP address) per Hunt Group for IP-to Tel calls. The IP-to-Tel call is forwarded to this IP destination (using 3xx response) if an unavailable FXO/FXS Hunt Group exists. This feature can be used, for example, to forward the IP-to-Tel call to another FXO/FXS device. The device forwards calls using this new table only if no alternative IPto-Tel routing has been configured or alternative routing fails, and the following call forward reason (included in the SIP Diversion header of 3xx messages) exists: "unavailable": All FXO/FXS lines pertaining to a Hunt Group are busy or unavailable The format of this parameter is as follows: [ForwardOnBusyTrunkDest] FORMAT ForwardOnBusyTrunkDest_Index = ForwardOnBusyTrunkDest_TrunkGroupId, ForwardOnBusyTrunkDest_ForwardDestination; [\ForwardOnBusyTrunkDest] Where: TrunkGroupId = Hunt Group for which you want to define a call forwarding destination. The default is 0. ForwardDestination = Alternative destination, using the syntax "host:port;transport=xxx"(i.e., IP address, port and transport type). For example, the below configuration forwards IP-to-Tel calls to destination IP address 10.13.4.12, port 5060 using transport protocol TCP, if Hunt Group ID 2 is busy: ForwardOnBusyTrunkDest 1 = 2, 10.13.4.12:5060;transport=tcp; Notes: • • The maximum number of indices (starting from Index 1) depends on the maximum number of Hunt Groups. For the destination, instead of a dotted-decimal IP address, FQDN can be used. Web: Tone Index Table [ToneIndex] This ini file table parameter configures the Tone Index table, which allows you to define Distinctive Ringing and Call Waiting tones per FXS endpoint(or a range of FXS endpoints), and based on calling number (source number prefix) for IP-to-Tel calls. Therefore, different tones can be played for an FXS endpoint, depending on the source number of the received call. The format of this parameter is as follows: [ToneIndex] FORMAT ToneIndex_Index = ToneIndex_FXSPort_First, ToneIndex_FXSPort_Last, ToneIndex_SourcePrefix, ToneIndex_PriorityIndex; [\ToneIndex] Version 6.0 27 February 2010 MP-11x & MP-124 Parameter Description Where, FXSPort_First = starting range of FXS ports. FXSPort_Last = end range of FXS ports. SourcePrefix = prefix of the calling number. PriorityIndex = index for Distinctive Ringing and Call Waiting tones (default is 0): Ringing tone index = Index in the CPT file for playing the ring tone. Call Waiting tone index = priority index plus FirstCallWaitingToneID(*). For example, if you want to select the Call Waiting tone defined in the CPT file at Index #9, then you can enter 1 as the priority index and the value 8 for FirstCallWaitingToneID. The summation of these values equals 9, i.e., index #9. For example, the configuration below plays the tone Index #3 to FXS ports 1 and 2 if the source number prefix of the received call is 20. ToneIndex 1 = 1, 2, 20*, 3; Notes: [SASEnableContactRepl ace] You can define up to 50 indices. This parameter is applicable only to FXS interfaces. Typically, the Ringing and/or Call Waiting tone played is indicated in the SIP Alert-info header field of the received INVITE message. If this header is not present, then the tone played is according to the settings in this table. For depicting a range of FXS ports, use the syntax x-y (e.g., "1-4" for ports 1 through 4). You can configure multiple entries for the same FXS port, with different source prefixes and tones. Enables the device to change the Contact header so that it points to the SAS host, and therefore, the top-most Via header and the Contact header point to the same host. [0] (default) = Disable - when relaying requests, the SAS agent adds a new SIP Via header (with the SAS IP address) as the top-most Via header and retains the original SIP Contact header. Thus, the topmost Via header and the Contact header point to different hosts. [1] = Enable - the device changes the Contact header so that it points to the SAS host and therefore, the top-most Via header and the Contact header point to the same host. Note: Operating in this mode causes all incoming dialog requests to traverse the SAS and thus, may cause load problems. Web: Enable RecordRoute [SASEnableRecordRout e] Determines whether the device's SAS application adds the SIP RecordRoute header to SIP requests. This ensures that SIP messages traverse the device's SAS agent, by including the SAS IP address in the Record-Route header. [0] Disable (default) [1] Enable The Record-Route header is inserted in a request by a SAS proxy to force future requests in the dialog session to be routed through the SAS agent. Each traversed proxy in the path can insert this header, causing SIP Release Notes 28 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 Parameter Description all future dialogs in the session to pass through it as well. When this feature is enabled, the SIP Record-Route header includes the URI "lr" parameter. The presence of this parameter indicates loose routing; the lack of 'lt' indicates strict routing. For example: Web: Add CIC [AddCicAsPrefix] Loose routing: Record-Route: <sip:server10.biloxi.com;lr> Strict routing: Record-Route: <sip:bigbox3.site3.atlanta.com> Determines whether to add the Carrier Identification Code (CIC) as a prefix to the destination phone number for IP-to-Tel calls. [0] No (default) [1] Yes When this parameter is enabled, the cic parameter in the incoming SIP INVITE can be used for IP-to-Tel routing decisions. It routes the call to the appropriate Hunt Group based on this parameter's value. For example: INVITE sip:5550001;[email protected]:5060;user=phone SIP/2.0 The destination number after number manipulation is cic+167895550001. Note: After the cic prefix is added, the IP-to-Trunk Group Routing table can be used to route this call to a specific Trunk Group. The Destination Number IP to Tel Manipulation table must be used to remove this prefix before placing the call to Tel. [UseBroadsoftDTG] Determines whether the device uses the “dtg” parameter for routing IPto-Tel calls to a specific Trunk Group. [0] Disable (default) [1] Enable When this parameter is enabled, if the Request URI in the received SIP INVITE includes the “dtg” parameter, the device routes the call to the Hunt Group according to its value. This parameter is used instead of the "tgrp/trunk-context" parameters. The "dtg" parameter appears in the INVITE Request URI (and in the To header). For example, the received SIP message below routes the call to Hunt Group ID 56: INVITE sip:[email protected];dtg=56;user=phone SIP/2.0 Note: If the Hunt Group is not found based on the "dtg" parameter, the IP to Trunk Group Routing table is used instead for routing the call to the appropriate Hunt Group. [SIPForceRport] Version 6.0 Determines whether the device sends SIP responses to the UDP port from where SIP requests are received even if the "rport" parameter is not included in the Via header. [0] (default) = Disabled - the device sends the SIP response to the UDP port defined in the Via header. If the Via header contains the "rport" parameter, the response is sent to the UDP port from where the SIP request is received. [1] = Enabled - SIP responses are sent to the UDP port from where SIP requests are received even if the "rport" parameter is not included in the Via header. 29 February 2010 MP-11x & MP-124 Parameter Web: Fake Retry After [sec] [FakeRetryAfter] Description Determines whether the device, upon receipt of a SIP 503 response without a Retry-After header, behaves as if the 503 response includes a Retry-After header and with the period (in seconds) specified by this parameter. [0] Disable Any positive value (in seconds) for enabling this feature When enabled, this feature allows the device to operate with proxy servers that do not include the Retry-After SIP header in SIP 503 (Service Unavailable) responses to indicate an unavailable service. The Retry-After header is used with the 503 (Service Unavailable) response to indicate how long the service is expected to be unavailable to the requesting SIP client. The device maintains a list of available proxies, by using the Keep-Alive (KA) mechanism. The device checks the availability of proxies by sending SIP OPTIONS every KA timeout to all proxies. If the device receives a SIP 503 response to an INVITE, it also marks that the proxy is out of service for the defined "Retry-After" period. Web: TGRP Routing Precedence [TGRProutingPrecedenc e] Determines the precedence method for routing IP-to-Tel calls according to the IP to Trunk Group Routing table or tgrp/dtg parameters. [0] (default) = IP-to-Tel routing is determined by the IP to Trunk Group Routing table (PSTNPrefix ini file parameter). If a matching rule is not found in this table, the device uses the Hunt Group parameters for routing the call. [1] = The device first places precedence on the tgrp/dtg parameters for IP-to-Tel routing. If the received INVITE request URI does not contain the tgrp/dtg parameters, or if the Hunt Group number is not defined, then the IP to Trunk Group Routing table is used for routing the call. Below is an example of an INVITE request URI with the tgrp parameter, indicating that the IP call should be routed to Hunt Group 7: INVITE sip:200;tgrp=7;[email protected];user=phone SIP/2.0 Note: For enabling routing based on the SIP tgrp parameter, the UseSIPTgrp parameter must be set to 2. Web: SIT Q850 Cause For NC [SITQ850CauseForNC] Determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-NC (No Circuit Found Special Information Tone) is detected from the Tel for IP-to-Tel calls. The valid range is 0 to 127. The default value is 34. Note: When not configured (i.e., default), the SITQ850Cause parameter is used. Web: SIT Q850 Cause For IC [SITQ850CauseForIC] Determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-IC (Operator Intercept Special Information Tone) is detected from the Tel for IP-to-Tel calls. The valid range is 0 to 127. The default value is -1 (not configured). Note: When not configured (i.e., default), the SITQ850Cause parameter is used. SIP Release Notes 30 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 Parameter Description Web: SIT Q850 Cause For VC [SITQ850CauseForVC] Determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-VC (Vacant Circuit - nonregistered number Special Information Tone) is detected from the Tel for IP-to-Tel calls. The valid range is 0 to 127. The default value is -1 (not configured). Note: When not configured (i.e., default), the SITQ850Cause parameter is used. Web: SIT Q850 Cause For RO [SITQ850CauseForRO] Determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when SIT-RO (Reorder - System Busy Special Information Tone) is detected from the Tel for IP-to-Tel calls. The valid range is 0 to 127. The default value is -1 (not configured). Note: When not configured (i.e., default), the SITQ850Cause parameter is used. [SelectSourceHeaderFor CalledNumber] [EnableRekeyAfter181] Determines the SIP header used for obtaining the called (destination) number (for IP-to-Tel calls). [0] Request-URI header (default) = Obtains the destination number from the user part of the Request-URI. [1] To header = Obtains the destination number from the user part of the To header. [2] P-Called-Party-ID header = Obtains the destination number from the P-Called-Party-ID header. Enables the device to send a Re-INVITE with a new (different) SRTP key (in the SDP) upon receipt of a SIP 181 response ("call is being forwarded"). [0] Disable (default) [1] Enable Note: This parameter is applicable only if SRTP is used. Version 6.0 31 February 2010 MP-11x & MP-124 Parameter [UnregistrationMode] Description Determines whether the device performs an explicit unregister. [0] Disable (default) [1] Enable = The device sends an asterisk (“*”) value in the Contact header, instructing the registrar server to remove all previous registration bindings. This parameter removes SIP UA registration bindings in a Registrar, according to RFC 3261. Registrations are soft state and expire unless refreshed, but can also be explicitly removed. A client can attempt to influence the expiration interval selected by the registrar. A UA requests the immediate removal of a binding by specifying an expiration interval of "0" for that contact address in a REGISTER request. UAs should support this mechanism so that bindings can be removed before their expiration interval has passed. Use of the "*" Contact header field value allows a registering UA to remove all bindings associated with an address-of-record (AOR) without knowing their precise values. Note: The REGISTER-specific Contact header field value of "*" applies to all registrations, but it can only be used if the Expires header field is present with a value of "0". [MaxBundleSyslogLengt h] The maximum size (in bytes) threshold of logged, bundled (into a single UDP packet) Syslog messages, after which they are sent to a Syslog server. The valid value range is 0 to 1220 (where 0 indicates that no bundling occurs). The default is 1220. Note: This parameter is applicable only if the GWDebugLevel parameter is set to 7. Web: Coders Table/Coder Group Settings [CodersGroup0] [CodersGroup1] [CodersGroup2] [CodersGroup3] [CodersGroup4] This ini file table parameter defines the device's coders. Up to five groups of coders can be defined, where each group can consist of up to 10 coders. The first Coder Group is the default coder list and the default Coder Group. These Coder Groups can later be assigned to IP or Tel Profiles. The format of this parameter is as follows: [ CodersGroup0] FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime, CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce; [ \CodersGroup0 ] Where, Index = Coder entry 0-9, i.e., up to 10 coders per group. Name = Coder name. Ptime = Packetization time (ptime) - how many coder payloads are combined into a single RTP packet. Rate = Packetization rate. PayloadType = Identifies the format of the RTP payload. Sce = Enables silence suppression: [0] Disabled (default) [1] Enabled For example, below are defined two Coder Groups (0 and 1): SIP Release Notes 32 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 Parameter Description [ CodersGroup0 ] FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime, CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce; CodersGroup0 0 = g711Alaw64k, 20, 0, 255, 0; CodersGroup0 1 = eg711Ulaw, 10, 0, 71, 0; CodersGroup0 2 = eg711Ulaw, 10, 0, 71, 0; [ \CodersGroup0 ] [ CodersGroup1 ] FORMAT CodersGroup1_Index = CodersGroup1_Name, CodersGroup1_pTime, CodersGroup1_rate, CodersGroup1_PayloadType, CodersGroup1_Sce; CodersGroup1 0 = Transparent, 20, 0, 56, 0; CodersGroup1 1 = g726, 20, 0, 23, 0; [ \CodersGroup1 ] The table below lists the supported coders: Version 6.0 Coder Name Packetization Time (msec) Rate (kbps) G.711 A-law [g711Alaw64k] 10, 20 (default), 30, 40, 50, 60, 80, 100, 120 Always 64 Always 8 Disable [0] Enable [1] G.711 U-law [g711Ulaw64k] 10, 20 (default), 30, 40, 50, 60, 80, 100, 120 Always 64 Always 0 Disable [0] Enable [1] EG.711 A-law [eg711Alaw] 10 (default), 20, 30 Always 64 Dynamic (96-127) N/A EG.711 U-law [eg711Ulaw] 10 (default), 20, 30 Always 64 Dynamic (96-127) N/A G.729 [g729] 10, 20 (default), 30, 40, 50, 60, 80, 100 Always 8 Always 18 Disable [0] Enable [1] Enable w/o Adaptations [2] G.722 [g722] 20 (default), 40, 60, 80, 100, 120 64 (default) Always 9 N/A G.723.1 [g7231] 30 (default), 60, 90 5.3 [0], 6.3 [1] (default) Always 4 Disable [0] Enable [1] G.726 [g726] 10, 20 (default), 30, 40, 50, 60, 80, 100, 120 16 [0], 24 [1], 32 [2] (default) 40 [3] Dynamic (0120) Disable [0] Enable [1] G.711Alaw_VBD [g711AlawVbd] 10, 20 (default), 30, 40, 50, 60, 80, 100, 120 Always 64 Dynamic (0120) N/A G.711Ulaw_VBD [g711UlawVbd] 10, 20 (default), 30, 40, 50, 60, 80, 100, 120 Always 64 Dynamic (0120) N/A T.38 [t38fax] N/A N/A N/A N/A 33 Payload Type Silence Suppression February 2010 MP-11x & MP-124 Parameter Description Notes: 1.8.2 The coder name is case-sensitive. Each coder type can appear only once per Coder Group. Only the packetization time of the first coder in the defined coder list is declared in INVITE/200 OK SDP, even if multiple coders are defined. The device always uses the packetization time requested by the remote side for sending RTP packets. If not specified, the packetization time is assigned the default value. The value of several fields is hard-coded according to common standards (e.g., payload type of G.711 U-law is always 0). Other values can be set dynamically. If no value is specified for a dynamic field, a default value is assigned. If a value is specified for a hardcoded field, the value is ignored. If silence suppression is not defined for a specific coder, the value defined by the parameter EnableSilenceCompression is used. If G.729 is selected and silence suppression is enabled (for this coder), the device includes the string 'annexb=no' in the SDP of the relevant SIP messages. If silence suppression is set to 'Enable w/o Adaptations', 'annexb=yes' is included. An exception is when the remote device is a Cisco gateway (IsCiscoSCEMode). Voice, RTP and RTCP Parameters The table below describes the new voice, RTP and RTCP parameters for Release 6.0. Table 1-2: New Voice, RTP and RTCP Parameters for Release 6.0 Parameter Description Web: FXS Coefficient Type Determines the FXS line characteristics (AC and DC) according to USA or Europe (TBR21) standards. [FXSCountryCoefficients] [66] Europe = TBR21 [70] USA = United States (default) Note: For this parameter to take effect, a device reset is required. SIP Release Notes 34 Document #: LTRT-65615 SIP Release Notes 1.8.3 1. What's New in Release 6.0 Networking Parameters The table below describes the new networking parameters for Release 6.0. Table 1-3: New Networking Parameters for Release 6.0 Parameter Description [DHCPRequestTFTPParams] Determines whether the device includes DHCP options 66 and 67 in DHCP Option 55 (Parameter Request List) for requesting the DHCP server for TFTP provisioning parameters. [0] = Disable (default) [1] = Enable Note: For this parameter to take effect, a device reset is required. Version 6.0 35 February 2010 MP-11x & MP-124 1.8.4 Security Parameters The table below describes the new security parameters for Release 6.0. Table 1-4: New Security Parameters for Release 6.0 Parameter Description Web: IP Security Associations Table [IPSecSATable] This ini file table parameter configures the IPSec SA table. This table allows you to configure the Internet Key Exchange (IKE) and IP Security (IPSec) protocols. You can define up to 20 IPSec peers. The format of this parameter is as follows: [ IPsecSATable ] FORMAT IPsecSATable_Index = IPsecSATable_RemoteEndpointAddressOrName, IPsecSATable_AuthenticationMethod, IPsecSATable_SharedKey, IPsecSATable_SourcePort, IPsecSATable_DestPort, IPsecSATable_Protocol, IPsecSATable_Phase1SaLifetimeInSec, IPsecSATable_Phase2SaLifetimeInSec, IPsecSATable_Phase2SaLifetimeInKB, IPsecSATable_DPDmode, IPsecSATable_IPsecMode, IPsecSATable_RemoteTunnelAddress, IPsecSATable_RemoteSubnetIPAddress, IPsecSATable_RemoteSubnetPrefixLength; [ \IPsecSATable ] Where: SIP Release Notes RemoteEndpointAddressOrName = IP address or DNS host name of the peer. AuthenticationMethod = Method used for peer authentication during IKE main mode. SharedKey = Defines the pre-shared key (in textual format). SourcePort = Defines the source port to which this configuration applies. DestPort = Defines the destination port to which this configuration applies. Protocol = Defines the protocol type to which this configuration applies. Standard IP protocol numbers should be used, e.g., 0 = Any protocol (default); 17 = UDP ;6 = TCP. InterfaceIndex = Interface Index. Phase1SaLifetimeInSec = Determines the duration (in seconds) for which the negotiated IKE SA (main mode) is valid. After the time expires, the SA is re-negotiated. Phase2SaLifetimeInSec = Determines the duration (in seconds) for which the negotiated IPSec SA (quick mode) is valid. After the time expires, the SA is re-negotiated. Phase2SaLifetimeInKB = Determines the maximum volume of traffic (in kilobytes) for which the negotiated IPSec SA (quick mode) is valid. DPDmode = Controls dead peer detection (DPD) according to RFC 3706. 36 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 Parameter Description IPsecMode = Selects the IPSec mode of operation: [0] = Transport mode (default) [1] = Tunnel mode RemoteTunnelAddress =IP address of the peer router. The default is 0.0.0.0. RemoteSubnetIPAddress = IP address of the remote subnetwork. The default is 0.0.0.0. RemoteSubnetPrefixLength = Prefix length of the Remote Subnet IP Address parameter (in bits). The default is 16. For example: IPsecSATable 1 = 0, 10.3.2.73, 0, 123456789, 0, 0, 0, 0, 28800, 3600; In the above example, a single IPSec/IKE peer (10.3.2.73) is configured. Pre-shared key authentication is selected, with the preshared key set to 123456789. In addition, a lifetime of 28800 seconds is selected for IKE and a lifetime of 3600 seconds is selected for IPSec. Notes: Each row in the table refers to a different IP destination. To support more than one Encryption / Authentication proposal, for each proposal specify the relevant parameters in the Format line. The proposal list must be contiguous. Web: IP Security Proposal Table [IPSecProposalTable] This ini file table parameter configures up to four IKE proposal settings, where each proposal defines an encryption algorithm, an authentication algorithm, and a Diffie-Hellman group identifier. [ IPsecProposalTable ] FORMAT IPsecProposalTable_Index = IPsecProposalTable_EncryptionAlgorithm, IPsecProposalTable_AuthenticationAlgorithm, IPsecProposalTable_DHGroup; [ \IPsecProposalTable ] Where: Version 6.0 EncryptionAlgorithm = Selects the encryption (privacy) algorithm: [0] NONE [1] DES CBC [2] 3DES CBC [3] AES (default) AuthenticationAlgorithm = Selects the message authentication (integrity) algorithm: [0] NONE [2] HMAC SHA1 96 [4] HMAC MD5 96 (default) DHGroup = Selects the Diffie-Hellman group: [0] Group 1 (768 Bits) [1] Group 2 (1024 Bits) - default 37 February 2010 MP-11x & MP-124 Parameter Description For example: IPsecProposalTable 0 = 3, 2, 1; IPsecProposalTable 1 = 2, 2, 1; In the example above, two proposals are defined: Proposal 0: AES, SHA1, DH group 2 Proposal 1: 3DES, SHA1, DH group 2 Notes: Web: 802.1x Mode [802.1xMode] Each row in the table refers to a different IKE peer. To support more than one Encryption / Authentication / DH Group proposal, for each proposal specify the relevant parameters in the Format line. The proposal list must be contiguous. Enables support for IEEE 802.1x physical port security. The device can function as an IEEE 802.1X supplicant. IEEE 802.1X is a standard for port-level security on secure Ethernet switches; when a unit is connected to a secure port, no traffic is allowed until the identity of the unit is authenticated. [0] Disabled (default) [1] EAP-MD5 [2] Protected EAP [3] EAP-TLS Web: 802.1x Username [802.1xUsername] Username for IEEE 802.1x support. The valid value is any string. The default is an empty string. Web: 802.1x Password [802.1xPassword] Password for IEEE 802.1x support. The valid value is any string. The default is an empty string. Web: 802.1x Verify Peer Certificate [802.1xVerifyPeerCertificate] Verify Peer Certificate for IEEE 802.1x support. SIP Release Notes [0] Disable (default) [1] Enable 38 Document #: LTRT-65615 SIP Release Notes 1.8.5 1. What's New in Release 6.0 Existing ini File Parameters Now Configurable in the Web The table below lists the ini file parameters that are now also configurable in the Web interface for Release 6.0. Table 1-5: ini File Parameters now Configurable in the Web Interface for Release 6.0 ini File Parameter [EnableDelayedOffer] Version 6.0 Web Parameter Description Enable Delayed Offer Determines whether the device sends the initial INVITE message with or without an SDP. Sending the first INVITE without SDP is typically done by clients for obtaining the farend's full list of capabilities before sending their own offer. (An alternative method for obtaining the list of supported capabilities is by using SIP OPTIONS, which is not supported by every SIP agent.) 39 [0] = The device sends the initial INVITE message with an SDP (default). [1] = The device sends the initial INVITE message without an SDP. February 2010 MP-11x & MP-124 1.9 Modified Parameters This section describes parameters from the previous release that have been modified in Release 6.0. These parameters can be configured using the ini file parameter (enclosed in square brackets) and/or the corresponding Web interface parameter (if supported). 1.9.1 SIP Parameters The table below describes SIP parameters from the previous release that have been modified in Release 6.0. Table 1-6: Modified SIP Parameters for Release 6.0 Parameter Web: SAS Survivability Mode [SASSurvivabilityMode] Web: Media Security Behavior [MediaSecurityBehaviour ] Description (Modification: New option 3, "Auto-answer REGISTER".) Determines the Survivability mode used by the SAS application. [0] Standard = All incoming INVITE and REGISTER requests are forwarded to the defined Proxy list in SASProxySet in Normal mode and handled by the SAS application in Emergency mode (default). [1] Always Emergency = The SAS application does not use KeepAlive messages towards the SASProxySet and instead, always operates in Emergency mode (as if no Proxy in the SASProxySet is available). [2] Ignore REGISTER = Use regular SAS Normal/Emergency logic (same as option 0), but when in Normal mode, incoming REGISTER requests are ignored. [3] Auto-answer REGISTER = When in Normal mode, the device responds to received REGISTER requests by sending a SIP 200 OK and enters the registrations in its SAS database. (Modification: New option "Preferable - Single Media".) Determines the device's mode of operation when SRTP is used (i.e., when EnableMediaSecurity is set to 1). [0] Preferable (default) = The device initiates encrypted calls. If negotiation of the cipher suite fails, an unencrypted call is established. Incoming calls that don't include encryption information are accepted. [1] Mandatory = The device initiates encrypted calls, but if negotiation of the cipher suite fails, the call is terminated. Incoming calls that don't include encryption information are rejected. [2] Preferable - Single Media = The device sends SDP with only a single media line ('m=') with RTP/AVP and crypto keys. If the remote SIP UA does not support SRTP, it ignores the crypto lines. Note: Before configuring this parameter, set the EnableMediaSecurity parameter to 1. Web: Tel Profile Settings Table [TelProfile] (Modification: New parameters Enable911PSAP and SwapTelToIpPhoneNumbers, EnableAGC, and ECNlpMode.) This ini file table parameter configures the Tel Profile table. Each Tel Profile ID includes a set of parameters, which are typically configured separately using their individual, "global" parameters. You can later SIP Release Notes 40 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 Parameter Description assign these Tel Profile IDs to other elements (e.g., to Trunk Groups TrunkGroup parameter). Therefore, Tel Profiles allows you to apply the same parameter settings of a group of parameters to multiple channels, and apply specific behaviours to specific channels. The format of this parameter is as follows: [TelProfile] FORMAT TelProfile_Index = TelProfile_ProfileName, TelProfile_TelPreference, TelProfile_CodersGroupID, TelProfile_IsFaxUsed, TelProfile_JitterBufMinDelay, TelProfile_JitterBufOptFactor, TelProfile_IPDiffServ, TelProfile_SigIPDiffServ, TelProfile_DtmfVolume, TelProfile_InputGain, TelProfile_VoiceVolume, TelProfile_EnableReversePolarity, TelProfile_EnableCurrentDisconnect, TelProfile_EnableDigitDelivery, TelProfile_EnableEC, TelProfile_MWIAnalog, TelProfile_MWIDisplay, TelProfile_FlashHookPeriod, TelProfile_EnableEarlyMedia, TelProfile_ProgressIndicator2IP, TelProfile_TimeForReorderTone, TelProfile_EnableDIDWink, TelProfile_IsTwoStageDial, TelProfile_DisconnectOnBusyTone, TelProfile_EnableVoiceMailDelay, TelProfile_DialPlanIndex, TelProfile_Enable911PSAP, TelProfile_SwapTelToIpPhoneNumbers, TelProfile_EnableAGC, TelProfile_ECNlpMode; [\TelProfile] Where, Enable911PSAP = Support for E911 DID protocol according to Bellcore GR-350-CORE standard: [-1] Not set (default) [0] Disabled [1] Enabled SwapTelToIpPhoneNumbers = Swaps the calling and called numbers received from the Tel side (for Tel-to-IP calls): [-1] Not set (default) [0] Disabled [1] Enabled EnableAGC = Activates the Automatic Gain Control (AGC) mechanism: [-1] Not set [0] Disabled [1] Enabled ECNlpMode = Defines the echo cancellation Non-Linear Processing (NLP) mode: [-1] Not set [0] Adaptive NLP [1] Disabled NLP [2] Silence Output NLP For example: TelProfile 1 = ITSP_audio, 1, 0, 0, 10, 10, 46, 40, -11, 0, 0, 0, 0, 0, 1, 0, 0, 700, 0, -1, 255, 0, 1, 1, 1, -1, 1, 0, 0, 0; Notes: Version 6.0 You can configure up to nine Tel Profiles (i.e., indices 1 through 9). The parameter TelPreference determines the priority of the Tel Profile (1 to 20, where 20 is the highest priority). If both IP and Tel 41 February 2010 MP-11x & MP-124 Parameter Description profiles apply to the same call, the coders and parameters common to Tel and IP Profiles of the preferred Profile are applied to that call. If the preference of the Tel and IP profiles is identical, the Tel Profile parameters take precedence. The parameter EnableVoiceMailDelay is applicable only if voice mail is enabled globally (using the parameter VoiceMailInterface). To use the settings of the corresponding global parameter, enter the value -1. For a detailed description of each parameter, refer to its corresponding "global" parameter. Web: IP Profile Settings Table [IPProfile] (Modification: New option 2 for MediaSecurityBehaviour.) This ini file table parameter configures the IP Profile table. Each IP Profile ID includes a set of parameters (which are typically configured separately using their individual, "global" parameters). You can later assign these IP Profiles to Tel-to-IP routing rules (Prefix parameter), IPto-Hunt Group routing rules (PSTNPrefix parameter), and IP Groups (IPGroup parameter). The format of this parameter is as follows: [IPProfile] FORMAT IPProfile_Index = IPProfile_ProfileName, IPProfile_IpPreference, IPProfile_CodersGroupID, IPProfile_IsFaxUsed, IPProfile_JitterBufMinDelay, IPProfile_JitterBufOptFactor, IPProfile_IPDiffServ, IPProfile_SigIPDiffServ, IpProfile_SCE, IPProfile_RTPRedundancyDepth, IPProfile_RemoteBaseUDPPort, IPProfile_CNGmode, IPProfile_VxxTransportType, IPProfile_NSEMode, IpProfile_IsDTMFUsed, IPProfile_PlayRBTone2IP, IPProfile_EnableEarlyMedia, IPProfile_ProgressIndicator2IP, IPProfile_EnableEchoCanceller, IPProfile_CopyDest2RedirectNumber,IPProfile_MediaSecurityBehaviou r, IPProfile_CallLimit, IPProfile_ DisconnectOnBrokenConnection, IPProfile_FirstTxDtmfOption, IPProfile_SecondTxDtmfOption, IPProfile_RxDTMFOption, IpProfile_EnableHold, IpProfile_InputGain, IpProfile_VoiceVolume, IpProfile_AddIEInSetup, IpProfile_SBCExtensionCodersGroupID, IPProfile_MediaIPVersionPreference, IPProfile_TranscodingMode; [\IPProfile] For example: IPProfile 0 = Sevilia, 1, 1, 0, 10, 10, 46, 40, 0, 0, 0, 0, 2, 0, 0, 0, 0, -1, 1, 0, 0, -1, 1, -1, -1, 1, 1, 0, 0, , -1, 4294967295, 0; Notes: SIP Release Notes You can configure up to nine IP Profiles (i.e., indices 1 through 9). The parameters SBCExtensionCodersGroupID, TranscodingMode AddIEInSetup, IsDTMFUsed (deprecated), and MediaIPVersionPreference are not applicable. The parameter IpPreference determines the priority of the IP Profile (1 to 20, where 20 is the highest preference). If both IP and Tel Profiles apply to the same call, the coders and common parameters (i.e., parameters configurable in both IP and Tel Profiles) of the preferred profile are applied to that call. If the Tel and IP Profiles are 42 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 Parameter Description identical, the Tel Profile parameters take precedence. Web: SIT Q850 Cause [SITQ850Cause] To assign the parameter's default value, enter two dollar signs ('$$'). To use the settings of the corresponding global parameter, enter the value -1. Configure intuitive names (ProfileName) for the IP Profiles so that they can later be easily identified. The parameter CallLimit defines the maximum number of concurrent calls allowed for that Profile. If the Profile is set to some limit, the device maintains the number of concurrent calls (incoming and outgoing) pertaining to the specific Profile. A limit value of [-1] indicates that there is no limitation on calls (default). A limit value of [0] indicates that all calls are rejected. When the number of concurrent calls is equal to the limit, the device rejects any new incoming and outgoing calls pertaining to that profile. RxDTMFOption configures the received DTMF negotiation method: [-1] not configured, use the global parameter; [0] don’t declare RFC 2833; [1] declare RFC 2833 payload type is SDP. FirstTxDtmfOption and SecondTxDtmfOption configures the transmit DTMF negotiation method: [-1] not configured, use the global parameter; for the remaining options, refer to the global parameter. IP Profiles can also be used when operating with a Proxy server (set the parameter AlwaysUseRouteTable to 1). For a detailed description of each parameter, refer to its corresponding global parameter. (Modification: Note added for additional SIT tones.) Determines the Q.850 cause value specified in the SIP Reason header that is included in a 4xx response when Special Information Tone (SIT) is detected on an IP-to-Tel call. The valid range is 0 to 127. The default value is 34. Note: For mapping specific SIT tones, you can use the SITQ850CauseForNC, SITQ850CauseForIC, SITQ850CauseForVC, SITQ850CauseForRO parameters. Web: Debug Level [GwDebugLevel] (Modification: New option 7 added.) Syslog debug logging level. [0] 0 (default) = Debug is disabled. [1] 1 = Flow debugging is enabled. [5] 5 = Flow, device interface, stack interface, session manager, and device interface expanded debugging are enabled. [7] 7 = The Syslog debug level automatically changes between level 5, level 1, and level 0, depending on the device's CPU consumption. Notes: Version 6.0 Usually set to 5 if debug traces are needed. Options 2, 3, 4, and 6 are not recommended for use. 43 February 2010 MP-11x & MP-124 Parameter Description Web: Proxy Set Table [ProxySet] (Modification: Maximum Proxy Sets increased from 6 to 10.) This ini file table parameter configures the Proxy Set ID table. It is used in conjunction with the ProxyIP ini file table parameter, which defines IP addresses per Proxy Set ID. The ProxySet ini file table parameter defines additional attributes per Proxy Set ID. This includes, for example, Proxy keep-alive and load balancing and redundancy mechanisms (if a Proxy Set contains more than one proxy address). The format of this parameter is as follows: [ProxySet] FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive, ProxySet_ProxyKeepAliveTime, ProxySet_ProxyLoadBalancingMethod, ProxySet_IsProxyHotSwap, ProxySet_SRD; [\ProxySet] For example: ProxySet 0 = 0, 60, 0, 0; ProxySet 1 = 1, 60, 1, 0; Notes: This table parameter can include up to 10 indices (0-9). For configuring IP addresses per Proxy Set ID, use the ini file parameter ProxyIP. The parameter ProxySet_SRD is not applicable. Web: Destination Phone Number Manipulation Table for Tel to IP Calls [NumberMapTel2IP] (Modification: Maximum entries increased from 100 to 120.) This ini file table parameter manipulates the destination number of Telto-IP calls. The format of this parameter is as follows: [NumberMapTel2Ip] FORMAT NumberMapTel2Ip_Index = NumberMapTel2Ip_DestinationPrefix, NumberMapTel2Ip_SourcePrefix, NumberMapTel2Ip_SourceAddress, NumberMapTel2Ip_NumberType, NumberMapTel2Ip_NumberPlan, NumberMapTel2Ip_RemoveFromLeft, NumberMapTel2Ip_RemoveFromRight, NumberMapTel2Ip_LeaveFromRight, NumberMapTel2Ip_Prefix2Add, NumberMapTel2Ip_Suffix2Add, NumberMapTel2Ip_IsPresentationRestricted, NumberMapTel2Ip_SrcTrunkGroupID, NumberMapTel2Ip_ SrcIPGroupID; [\NumberMapTel2Ip] For example: NumberMapTel2Ip 0 = 01,$$,*,0,0,2,$$,$$,971,$$,$$,$$,$$; NumberMapTel2Ip 1 = 10,10,*,255,255,3,0,5,100,$$,255,$$,$$; Notes: SIP Release Notes This table parameter can include up to 120 indices (0-119). The parameters SourceAddress and IsPresentationRestricted are not applicable. Set these to $$. 44 Document #: LTRT-65615 SIP Release Notes 1. What's New in Release 6.0 Parameter 1.9.2 Description The parameters NumberMapTel2Ip_ SrcIPGroupID, NumberMapTel2Ip_NumberType and NumberMapTel2Ip_NumberPlan are not applicable. Set these to $$. The parameter RemoveFromLeft, RemoveFromRight, Prefix2Add, Suffix2Add, LeaveFromRight, NumberType, and NumberPlan are applied if the called and calling numbers match the DestinationPrefix and SourcePrefix conditions. The manipulation rules are executed in the following order: RemoveFromLeft, RemoveFromRight, LeaveFromRight, Prefix2Add, and Suffix2Add. Parameters can be skipped by using two dollar signs ('$$'). Voice, RTP and RTCP Parameters The table below describes voice, RTP and RTCP parameters from the previous release that have been modified in Release 6.0. Table 1-7: Modified Voice, RTP and RTCP Parameter for Release 6.0 Parameter Web: FXO Coefficient Type [CountryCoefficients] Description (Modification: Web support.) Determines the FXO line characteristics (AC and DC) according to USA or Europe (TBR21) standards. [66] Europe = TBR21 [70] USA = United States (default) Note: For this parameter to take effect, a device reset is required. Web: Fax Relay Max Rate (bps) [FaxRelayMaxRate] (Modification: Additional values 6 to 13.) Maximum rate (in bps) at which fax relay messages are transmitted (outgoing calls). • • • • • • • • • • • • • • [0] 2400bps = 2.4 kbps [1] 4800bps = 4.8 kbps [2] 7200bps = 7.2 kbps [3] 9600bps = 9.6 kbps [4] 12000bps = 12.0 kbps [5] 14400bps = 14.4 kbps (default) [6] 16800bps = 16.8 kbps [7] 19200bps = 19.2 kbps [8] 21600bps = 21.6 kbps [9] 24000bps = 24 kbps [10] 26400bps = 26.4 kbps [11] 28800bps = 28.8 kbps [12] 31200bps = 31.2 kbps [13] 33600bps = 33.6 kbps Note: The rate is negotiated between the sides (i.e., the device adapts to the capabilities of the remote side). Version 6.0 45 February 2010 MP-11x & MP-124 1.10 Obsolete Parameters The table below lists parameters from the previous release that are now obsolete in Release 6.0. Table 1-8: Obsolete Parameters Parameter Description [IPSec_SPD_Table] This parameter is now obsolete and has been replaced by the new ini file parameter tables IPsecSATable and IPsecProposalTable. [IPSec_IKEDB_Table] This parameter is now obsolete and has been replaced by the new ini file parameter tables IPsecSATable and IPsecProposalTable. [CoderName] This parameter is now obsolete and has been replaced by the new ini file parameter CodersGroup. [IsUseToHeaderAsCalledNumber] This parameter is now obsolete and has been replaced by the new ini file parameter SelectSourceHeaderForCalledNumber. [FXSCoefFilename] This parameter is now obsolete and has been replaced by the new ini file parameter FXSCountryCoefficients. [FXSLoopCharacteristicsFilename] This parameter has been replaced by the new ini file parameter FXSCountryCoefficients. [FXSCoeffFileURL] This parameter is now obsolete. [FXOLoopCharacteristicsFilename] This parameter is now obsolete and has been replaced by the existing ini file parameter CountryCoefficients. [FXOCoefFilename] This parameter is now obsolete and has been replaced by the existing ini file parameter CountryCoefficients. SIP Release Notes 46 Document #: LTRT-65615 SIP Release Notes 2. Supported Features 2 Supported Features 2.1 SIP Features 2.1.1 Supported SIP Features The device supports the following main SIP features: Reliable User Datagram Protocol (UDP) transport, with retransmissions. Transmission Control Protocol (TCP) Transport layer. SIPS using TLS. T.38 real time Fax (using SIP). Note: If the remote side includes the fax maximum rate parameter in the SDP body of the INVITE message, the device returns the same rate in the response SDP. Operates with Proxy or without Proxy, using an internal routing table. Fallback to internal routing table if Proxy is not responding. Supports up to 15 Proxy servers. If the primary Proxy fails, the device automatically switches to a redundant Proxy. Supports domain name resolving using DNS NAPTR and SRV records for Proxy, Registrar and domain names that appear in the Contact and Record-Route headers. Supports Load Balancing over Proxy servers using Round Robin or Random Weights. Proxy or Registrar Registration, such as: REGISTER sip:servername SIP/2.0 VIA: SIP/2.0/UDP 212.179.22.229;branch=z9hG4bRaC7AU234 From: <sip:GWRegistrationName@sipgatewayname>;tag=1c29347 To: <sip:GWRegistrationName@sipgatewayname> Call-ID: [email protected] Seq: 1 REGISTER Expires: 3600 Contact: sip:[email protected] Content-Length: 0 The "servername" string is defined according to the following rules: Version 6.0 • The "servername" is equal to "RegistrarName" if configured. The "RegistrarName" can be any string. • Otherwise, the "servername" is equal to "RegistrarIP" (either FQDN or numerical IP address), if configured. • Otherwise the "servername" is equal to "ProxyName" if configured. The "ProxyName" can be any string. • Otherwise the "servername" is equal to "ProxyIP" (either FQDN or numerical IP address). 47 February 2010 MP-11x & MP-124 The parameter GWRegistrationName can be any string. This parameter is used only if registration is Per Gateway. If the parameter is not defined, the parameter UserName is used instead. If the registration is per endpoint, the endpoint phone number is used. The 'sipgatewayname' parameter (defined in the ini file or set from the Web browser), can be any string. Some Proxy servers require that the 'sipgatewayname' (in REGISTER messages) is set equal to the Registrar / Proxy IP address or to the Registrar / Proxy domain name. The 'sipgatewayname' parameter can be overwritten by the TrunkGroupSettings_GatewayName value if the TrunkGroupSettings_RegistrationMode is set to “Per Endpoint”. REGISTER messages are sent to the Registrar's IP address (if configured) or to the Proxy's IP address. A single message is sent once per device, or messages are sent per channel according to the parameter AuthenticationMode. There is also an option to configure registration mode per Trunk Group using the TrunkGroupSettings table. The registration request is resent according to the parameter RegistrationTimeDivider. For example, if RegistrationTimeDivider = 70 (%) and Registration Expires time = 3600, the device resends its registration request after 3600 x 70% = 2520 sec. The default value of RegistrationTimeDivider is 50%. If registration per channel is selected, on device startup, the device sends REGISTER requests according to the maximum number of allowed SIP dialogs (configured by the parameter NumberOfActiveDialogs). After each received response, the subsequent REGISTER request is sent. Proxy and Registrar Authentication (handling 401 and 407 responses) using Digest method. Accepted challenges are kept for future requests to reduce the network traffic. Single device Registration or multiple Registration of all device endpoints. Supported methods: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, INFO, REFER, UPDATE, NOTIFY, PRACK, SUBSCRIBE and PUBLISH. Modifying connection parameters for an already established call (re-INVITE). Working with Redirect server and handling 3xx responses. Early media (supporting 183 Session Progress). PRACK reliable provisional responses (RFC 3262). Call Hold and Transfer Supplementary services using REFER, Refer-To, Referred-By, Replaces and NOTIFY messages. Supports RFC 3711, Secured RTP and Key Exchange, according to RFC 4568. Supports RFC 3489, Simple Traversal of UDP Through NATs (STUN). Supports RFC 3327, Adding 'Path' to Supported header. Supports RFC 3581, Symmetric Response Routing. Supports RFC 3605, RTCP Attribute in SDP. Supports RFC 3326, Reason header. Supports RFC 4028, Session Timers in SIP. Supports network asserted identity and privacy (RFC 3325 and RFC 3323). Support RFC 3903, SIP Extension for Event State Publication. Support RFC 3953, The Early Disposition Type for SIP. Support for RFC 3966, The tel URI for Telephone Numbers. Support RFC 4244, An Extension to SIP for Request History Information. SIP Release Notes 48 Document #: LTRT-65615 SIP Release Notes 2. Supported Features Supports Tel URI (Uniform Resource Identifier) according to RFC 2806 bis. Supports ITU V.152 - Procedures for supporting Voice-Band Data over IP Networks. Remote party ID <draft-ietf-sip-privacy-04.txt>. Supports obtaining Proxy Domain Name(s) from DHCP (Dynamic Host Control Protocol) according to RFC 3361. Supports handling forking proxy multiple responses. RFC 2833 Relay for DTMF Digits, including payload type negotiation. DTMF out-of-band transfer using: • INFO method <draft-choudhuri-sip-info-digit-00.txt> • INFO method, compatible with Cisco gateways • NOTIFY method <draft-mahy-sipping-signaled-digits-01.txt> • INFO method, compatible with Korea Telecom format SIP URL: sip:”phone number”@IP address (such as [email protected], where “122556” is the phone number of the source or destination) or sip:”phone_number”@”domain name”, such as [email protected]. Note that the SIP URI host name can be configured differently per called number. Supports RFC 4040, RTP payload format for a 64 kbit/s transparent data. Can negotiate coder from a list of given coders. Supports negotiation of dynamic payload types. Supports multiple ptime values per coder. Supports RFC 3389, RTP Payload for Comfort Noise. Supports RFC 3824, Using E.164 numbers with SIP (ENUM). Supports receipt and DNS resolution of FQDNs received in SDP. Supports <draft-ietf-sip-gruu-09>, Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in SIP Responds to OPTIONS messages both outside a SIP dialog and in mid-call. Generates SIP OPTIONS messages as Proxy keep-alive mechanism. Publishes the total number of free Tel channels in a 200 OK response to an OPTIONS requests. Support RFC 3310, HTTP Digest Authentication Using Authentication and Key Agreement (AKA). Supports receipt of a REFER method outside of a dialog. Support RFC 4458, SIP URIs for Applications such as voice mail and Interactive Voice Response (IVR). Support RFC 3608, SIP Extension Header Field for Service Route Discovery During Registration. Support RFC 3911, The SIP Join Header (Partial). Support RFC 4730, A SIP Event Package for Key Press Stimulus (KPML) (Partial). Version 6.0 49 February 2010 MP-11x & MP-124 2.1.2 Support RFC 3455, Private Header (P-Header) Extensions to SIP for the 3rdGeneration Partnership Project (3GPP) [Partial]. Support RFC 4235, An INVITE-Initiated Dialog Event Package for SIP [Partial]. Support RFC 3680, A SIP Event Package for Registrations. Unsupported SIP Features The following SIP features are not supported: 2.1.3 MESSAGE method Preconditions (RFC 3312) SDP - Simple Capability Declaration (RFC 3407) S/MIME SIP Compliance Tables The SIP device complies with RFC 3261, as shown in the following subsections. 2.1.3.1 SIP Functions The device supports the following SIP Functions: Table 2-1: Supported SIP Functions Function Supported User Agent Client (UAC) Yes User Agent Server (UAS) Yes Proxy Server The device supports working with third-party Proxy Servers such as Nortel CS1K/CS2K, Avaya, Microsoft OCS, Alcatel, 3Com, BroadSoft, Snom, Cisco and many others Redirect Server The device supports working with third-party Redirection servers Registrar Server The device supports working with third-party Registration servers Proxy Server Third party, only tested with, amongst others, Ubiquity, Delta3, Microsoft, 3Com, BroadSoft, Snom and Cisco Proxies SIP Release Notes 50 Document #: LTRT-65615 SIP Release Notes 2.1.3.2 2. Supported Features SIP Methods The device supports the following SIP Methods: Table 2-2: Supported SIP Methods Method Supported Comments INVITE Yes ACK Yes BYE Yes CANCEL Yes REGISTER Yes Send only REFER Yes Inside and outside of a dialog NOTIFY Yes INFO Yes OPTIONS Yes PRACK Yes UPDATE Yes PUBLISH Yes SUBSCRIBE Yes 2.1.3.3 Send only SIP Headers The device supports the following SIP Headers: Table 2-3: Supported SIP Headers Header Field Supported Accept Yes Accept–Encoding Yes Alert-Info Yes Allow Yes Also Yes Asserted-Identity Yes Authorization Yes Call-ID Yes Call-Info Yes Contact Yes Content-Disposition Yes Content-Encoding Yes Version 6.0 51 February 2010 MP-11x & MP-124 Header Field Supported Content-Length Yes Content-Type Yes Cseq Yes Date Yes Diversion Yes Encryption No Expires Yes Fax Yes From Yes History-Info Yes Join Yes Max-Forwards Yes Messages-Waiting Yes MIN-SE Yes Organization No P-Associated-URI Yes (Receive Only) P-Asserted-Identity Yes P-Charging-Vector Yes P-Preferred-Identity Yes Priority Yes Proxy- Authenticate Yes Proxy- Authorization Yes Proxy- Require Yes Prack Yes Reason Yes Record- Route Yes Refer-To Yes Referred-By Yes Replaces Yes Require Yes Remote-Party-ID Yes Response- Key Yes Retry-After Yes Route Yes Rseq Yes Session-Expires Yes SIP Release Notes 52 Document #: LTRT-65615 SIP Release Notes 2. Supported Features Header Field Supported Server Yes Service-Route Yes SIP-If-Match Yes Subject Yes Supported Yes Target-Dialog Yes Timestamp Yes To Yes Unsupported Yes User- Agent Yes Via Yes Voicemail Yes Warning Yes WWW- Authenticate Yes 2.1.3.4 SDP Headers The device supports the following SDP Headers: Table 2-4: Supported SDP Headers SDP Header Element Supported v - Protocol version Yes o - Owner/ creator and session identifier Yes a - Attribute information Yes c - Connection information Yes d - Digit Yes m - Media name and transport address Yes s - Session information Yes t - Time alive header Yes b - Bandwidth header Yes u - Uri Description Header Yes e - Email Address header Yes i - Session Info Header Yes p - Phone number header Yes y - Year Yes Version 6.0 53 February 2010 MP-11x & MP-124 2.1.3.5 SIP Responses The device supports the following SIP responses: 1xx Response - Information Responses 2xx Response - Successful Responses 3xx Response - Redirection Responses 4xx Response - Client Failure Responses 5xx Response - Server Failure Responses 6xx Response - Global Responses 2.1.3.5.1 1xx Response – Information Responses Table 2-5: Supported 1xx SIP Responses 1xx Response Supported Comments 100 Trying Yes The device generates this response upon receiving a Proceeding message from ISDN or immediately after placing a call for CAS signaling. 180 Ringing Yes The device generates this response for an incoming INVITE message. Upon receiving this response, the device waits for a 200 OK response. 181 Call is Being Forwarded Yes The device doesn't generate these responses. However, the device does receive them. The device processes these responses the same way that it processes the 100 Trying response. 182 Queued Yes The device generates this response in Call Waiting service. When the SIP device receives a 182 response, it plays a special waiting Ringback tone to the telephone side. 183 Session Progress Yes The device generates this response if the Early Media feature is enabled and if the device plays a Ringback tone to IP 2.1.3.5.2 2xx Response – Successful Responses Table 2-6: Supported 2xx SIP Responses 2xx Response Supported Comments 200 OK Yes - 202 Accepted Yes - SIP Release Notes 54 Document #: LTRT-65615 SIP Release Notes 2. Supported Features 2.1.3.5.3 3xx Response – Redirection Responses Table 2-7: Supported 3xx SIP Responses 3xx Response Supported Comments 300 Multiple Choice Yes The device responds with an ACK, and then resends the request to the first new address in the contact list. 301 Moved Permanently Yes The device responds with an ACK, and then resends the request to the new address. 302 Moved Temporarily Yes The device generates this response when call forward is used to redirect the call to another destination. If such a response is received, the calling device initiates an INVITE message to the new destination. 305 Use Proxy Yes The device responds with an ACK, and then resends the request to a new address. 380 Alternate Service Yes The device responds with an ACK, and then resends the request to a new address. 2.1.3.5.4 4xx Response – Client Failure Responses Table 2-8: Supported 4xx SIP Responses 4xx Response Supported Comments 400 Bad Request Yes The device doesn't generate this response. Upon receipt of this message, and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 401 Unauthorized Yes Authentication support for Basic and Digest. Upon receiving this message, the device issues a new request according to the scheme received on this response. 402 Payment Required Yes The device doesn't generate this response. Upon receipt of this message, and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 403 Forbidden Yes The device doesn't generate this response. Upon receipt of this message, and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 404 Not Found Yes The device generates this response if it is unable to locate the callee. Upon receiving this response, the device notifies the User with a Reorder Tone. 405 Method Not Allowed Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 406 Not Acceptable Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 407 Proxy Authentication Required Yes Authentication support for Basic and Digest. Upon receiving this message, the device issues a new request according to the scheme received on this response. Version 6.0 55 February 2010 MP-11x & MP-124 4xx Response Supported Comments 408 Request Timeout Yes The device generates this response if the no-answer timer expires. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 409 Conflict Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 410 Gone Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 411 Length Required Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 413 Request Entity Too Large Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 415 Unsupported Media Yes If the device receives a 415 Unsupported Media response, it notifies the User with a Reorder Tone. The device generates this response in case of SDP mismatch. 420 Bad Extension Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 423 Interval Too Brief Yes The device does not generate this response. On reception of this message the device uses the value received in the Min-Expires header as the registration time. 433 Anonymity Disallowed Yes If the device receives a 433 Anonymity Disallowed, it sends a DISCONNECT message to the PSTN with a cause value of 21 (Call Rejected). In addition, the device can be configured, using the Release Reason Mapping, to generate a 433 response when any cause is received from the PSTN side. 480 Temporarily Unavailable Yes If the device receives a 480 Temporarily Unavailable response, it notifies the User with a Reorder Tone. This response is issued if there is no response from remote. 481 Call Leg/Transaction Does Not Exist Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 482 Loop Detected Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 483 Too Many Hops Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 484 Address Incomplete Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. SIP Release Notes 56 Document #: LTRT-65615 SIP Release Notes 2. Supported Features 4xx Response Supported Comments 485 Ambiguous Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 486 Busy Here Yes The SIP device generates this response if the called party is off-hook and the call cannot be presented as a call waiting call. Upon receipt of this response, the device notifies the User and generates a busy tone. 487 Request Canceled Yes This response indicates that the initial request is terminated with a BYE or CANCEL request. 488 Not Acceptable Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call. 491 Request Pending Yes When acting as a UAS: the device sent a re-INVITE on an established session and is still in progress. If it receives a re-INVITE on the same dialog, it returns a 491 response to the received INVITE. When acting as a UAC: If the device receives a 491 response to a re-INVITE, it starts a timer. After the timer expires, the UAC tries to send the re-INVITE again. 2.1.3.5.5 5xx Response – Server Failure Responses Table 2-9: Supported 5xx SIP Responses 5xx Response 500 Internal Server Error 501 Not Implemented 502 Bad gateway 503 Service Unavailable 504 Gateway Timeout 505 Version Not Supported Comments Upon receipt of any of these Responses, the device releases the call, sending an appropriate release cause to the PSTN side. The device generates a 5xx response according to the PSTN release cause coming from the PSTN. 2.1.3.5.6 6xx Response – Global Responses Table 2-10: Supported 6xx SIP Responses 6xx Response 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable Version 6.0 Comments Upon receipt of any of these Responses, the device releases the call, sending an appropriate release cause to the PSTN side. 57 February 2010 MP-11x & MP-124 2.2 DSP Firmware Templates The device supports the following DSP firmware templates: Table 2-11: DSP Firmware Template for MediaPack Series DSP Template 0 1 Number of Channels Default SRTP Enabled Default SRTP Enabled MP-112 FXS/FXO 2 2 2 2 MP-114 FXS/FXO 4 3 3 3 MP-118 FXS/FXO 8 6 6 6 MP-124 24 20 20 20 Voice Coder G.711 A/Mu-law PCM Yes Yes G.726 ADPCM Yes Yes G.727 ADPCM Yes Yes G.723.1 Yes Yes G.729 A, B Yes Yes EG.711 Yes - - Yes G.722 Notes: SIP Release Notes • Installation and use of vocoders is subject to obtaining the appropriate license and to royalty payments. • The number of channels refers to the device's maximum channel capacity. • For other DSP template configurations, please contact AudioCodes. 58 Document #: LTRT-65615 SIP Release Notes 3 3. Known Constraints Known Constraints This section lists known constraints in Release 6.0. Note: Due to the improved ini file format for tables, it's not possible to load an ini file that was used by a device running software version 5.2 or later to a device using an earlier version (e.g. 5.0). This can result in an invalid configuration. For additional information, contact AudioCodes. 3.1 Voice, RTP and RTCP Constraints This release includes the following known voice, RTP and RTCP constraints: 3.2 1. RFC 2198 Redundancy mode with RFC 2833 is not supported (i.e., if a complete DTMF digit is lost, it is not reconstructed). The current RFC 2833 implementation supports Redundancy for lost inter-digit information. Since the channel can construct the entire digit from a single RFC 2833 end packet, the probability of such inter-digit information loss is very low. 2. When using SRTP, the number of basic codec frames per RTP packet cannot be greater than one. In addition, the RTP Redundancy (RFC 2198) feature cannot be activated. 3. When using a coder sample interval of 5 or 10 msec, the channel capacity may be reduced. 4. The duration resolution of the On and Off time digits when dialing to the network using RFC 2833 relay is dependent on the basic frame size of the coder being used. Infrastructure Constraints This release includes the following known infrastructure constraints: 1. Version 6.0 The following parameters do not return to their default values when attempting to restore them to defaults using the Web or SNMP interfaces, or when loading a new ini file using BootP/TFTP: • VLANMode • VLANNativeVLANID • RoutingTableDestinationsColumn • RoutingTableDestinationPrefixLensColumn • RoutingTableInterfacesColumn • RoutingTableGatewaysColumn • RoutingTableHopsCountColumn • RoutingTableDestinationMasksColumn • EnableDHCPLeaseRenewal • RoutingTableDestinationMasksColumn • IPSecMode • CASProtocolEnable 59 February 2010 MP-11x & MP-124 3.3 • EnableSecureStartup • UseRProductName • LogoWidth • WebLogoText • UseWeblogo • UseProductName 2. The Multiple Interface table does not return to default values when attempting to restore it to defaults using the Web or SNMP interfaces, or when loading a new ini file using BootP/TFTP. 3. Files loaded to the device must not contain spaces in their file name. Including spaces in the name prevents the file from being saved to the device's flash memory. Networking Constraints This release includes the following known networking constraints: 3.4 1. In certain cases, when the Spanning-Tree algorithm is enabled on the external Ethernet switch port that is connected to the device, the external switch blocks all traffic from entering and leaving the device for some time after the device is reset. This may result in the loss of important packets such as BootP and TFTP requests, which in turn, may cause a failure in device start-up. A possible workaround is to set the ini file parameter BootPRetries to 5, causing the device to issue 20 BootP requests for 60 seconds. Another workaround is to disable the spanning tree on the port of the external switch that is connected to the device. 2. Configuring the device to auto-negotiate mode while the opposite port is set manually to full-duplex (either 10BaseT or 100BaseTX) is invalid. It is also invalid to set the device to one of the manual modes while the opposite port is configured differently. The user is encouraged to always prefer full-duplex connections over half-duplex, and 100BaseTX over 10BaseT (due to the larger bandwidth). 3. PPPoE is not supported. 4. Debug Recording: • Only one IP target is allowed. • Maximum of 50 trace rules are allowed simultaneously. • Maximum of 5 media stream recordings are allowed simultaneously. Security Constraints This release includes the following known security constraint: 1. The value of the Active IPSec SAS Performance Monitoring element is not supported. SIP Release Notes 60 Document #: LTRT-65615 SIP Release Notes 3.5 3. Known Constraints Web Constraints This release includes the following known Web constraints: 3.6 1. The fax counters (Attempted Fax Calls Counter and Successful Fax Calls Counter) in the 'Status & Diagnostics' page do not function correctly. 2. If the Home button is clicked when the Scenario mode is active, the Scenario mode is not exited. 3. Scrolling errors appear on the Home page when reducing the size of the browser's window (i.e. window not maximized). 4. On the 'Software Upgrade Wizard' page, the software upgrade process must be completed prior to clicking the Back button. Clicking the Back button before the wizard completes causes a display distortion. 5. On the 'IP Interface Status' page (under the Status & Diagnostics menu), the IP addresses may not be fully displayed if the address is greater than 25 characters. 6. On the 'IP Settings' page, adding an interface with invalid characters (e.g., <, >, ", and ') may result in a corrupted Web page. Submitting the corrupted Web page may result in an unexpected behavior such as no response from the device. 7. The parameter FlashHookPeriod can be configured only per device and not per FXS or FXO port. SNMP Constraints This release includes the following known Simple Network Management Protocol (SNMP) constraints: 3.7 1. When configuring acSysInterfaceTable using SNMP or the Web interface, validation is performed only after device reset. 2. When defining or deleting SNMPv3 users, the v3 trap user must not be the first to be defined or the last to be deleted. If there are no non-default v2c users, this results in a loss of SNMP contact with the device. CLI Constraints This release includes the following known command-line interface (CLI) constraints: 1. Version 6.0 When connecting to the device using Telnet (CLI), Syslog messages do not appear by default. The Show Log command can be used to enable this feature. 61 February 2010 MP-11x & MP-124 Reader’s Notes SIP Release Notes 62 Document #: LTRT-65615 SIP Release Notes 4 4. Resolved Constraints Resolved Constraints This section lists constraints from previous releases that have been resolved in Release 6.0. 4.1 Voice, RTP and RTCP Resolved Constraints The following voice, RTP and RTCP constraints from previous releases have now been resolved in Release 6.0: 4.2 1. Setting the parameter V.21 Transport Type to “Bypass” and the Fax Transport Type to “Relay” results in entering the Fax Relay mode at the 2,100 Hz signal. Only at the end of this signal does the channel enter “Bypass” mode if V.21 Modem is detected. To avoid this, the use should either switch the Fax setting to “Bypass” or the V.21 setting to “Transparent”. 2. The number of RTP payloads packed in a single G.729 packet (M channel parameter) is limited to 5. Web Resolved Constraints The following Web constraints from previous releases have now been resolved in Release 6.0: 1. 4.3 If the Home button is clicked when a device is loaded in Scenario mode, the Scenario mode is not closed. SNMP Resolved Constraints The following SNMP constraints from previous releases have now been resolved in Release 6.0: 1. SNMP configuration for the Access List table is not activated when using CreateAndGo for the row status MIB object. Instead, CreateAndWait followed by Active should be used. 2. Action Result for the following file types, describes the file download to the blade but not the related application’s parsing results. The file types are: Version 6.0 • Voice prompt • Xml • User info 63 February 2010 MP-11x & MP-124 Reader’s Notes SIP Release Notes 64 Document #: LTRT-65615 SIP Release Notes 5 5. Earlier Releases Earlier Releases Details of previous releases can be found in the Release Notes of Version 5.8, published by AudioCodes on 18 June 2009. Version 6.0 65 February 2010 Release Notes Version 6.0 www.audiocodes.com
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