LTRT-65610 MP-11x and MP-124 SIP Release Notes Ver 5.4

Release Notes
Version 5.4
Document #: LTRT-65610
May 2008
SIP Release Notes
Contents
Table of Contents
1 What's New in Release 5.4..................................................................................7
1.1 Supported Hardware Platforms................................................................................ 7
1.1.1 1.1.2 1.1.3 New Products Introduced in this Release..................................................................7 Support of the Existing Hardware Platforms..............................................................7 Hardware Platforms No Longer Supported ...............................................................7
1.2 1.3 1.4 1.5 1.6 1.7
General Gateway New Features.............................................................................. 8 SIP New Features .................................................................................................. 14 Web and SNMP New Features .............................................................................. 17 New Parameters .................................................................................................... 18 Modified Parameters .............................................................................................. 28 Obsolete Parameters ............................................................................................. 38
Supported Features ..........................................................................................39

2.1 2.2 SIP Features .......................................................................................................... 39
2.1.1 2.1.2 2.2.1 2.2.2 2.2.3 2.2.4 2.2.5 Supported SIP Features ..........................................................................................39 Unsupported SIP Features ......................................................................................42 SIP Functions ..........................................................................................................42 SIP Methods ............................................................................................................42 SIP Headers ............................................................................................................43 SDP Headers...........................................................................................................45 SIP Responses ........................................................................................................45 2.2.5.1 1xx Response Information Responses ................................................ 46 2.2.5.2 2xx Response Successful Responses ................................................. 46 2.2.5.3 3xx Response Redirection Responses ................................................ 46 2.2.5.4 4xx Response Client Failure Responses ............................................. 47 2.2.5.5 5xx Response Server Failure Responses............................................ 49 2.2.5.6 6xx Response Global Responses ........................................................ 49
SIP Compliance Tables.......................................................................................... 42
Known Constraints ...........................................................................................51

3.1 3.2 3.3 3.4 3.5 SIP Constraints ...................................................................................................... 51 Gateway Constraints.............................................................................................. 51 Web Constraints..................................................................................................... 53 SNMP Constraints.................................................................................................. 54 CLI Constraints ...................................................................................................... 54
4 5
Resolved Constraints .......................................................................................55 Earlier Releases ................................................................................................57
Version 5.4
May 2008
MediaPack Series
List of Tables
Table 1-1: Release 5.4 New Web / [ini] File Parameters .......................................................................18 Table 1-2: Release 5.4 Modified Web / [ini] File Parameters.................................................................28 Table 1-3: Release 5.4 Obsolete Web / [ini ] File Parameters ...............................................................38 Table 2-1: Supported SIP Functions ......................................................................................................42 Table 2-2: Supported SIP Methods ........................................................................................................42 Table 2-3: Supported SIP Headers ........................................................................................................43 Table 2-4: Supported SDP Headers.......................................................................................................45 Table 2-5: Supported 1xx SIP Responses .............................................................................................46 Table 2-6: Supported 2xx SIP Responses .............................................................................................46 Table 2-7: Supported 3xx SIP Responses .............................................................................................46 Table 2-8: Supported 4xx SIP Responses .............................................................................................47 Table 2-9: Supported 5xx SIP Responses .............................................................................................49 Table 2-10: Supported 6xx SIP Responses ...........................................................................................49
SIP Release Notes
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) SIP 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. Copyright 2008 AudioCodes Ltd. All rights reserved. This document is subject to change without notice. Date Published: May-20-2008 Date Printed: May-21-2008
Tip:
When viewing this manual on CD, Web site or on any other electronic copy, all cross-references are hyperlinked. Click on the page or section numbers (shown in blue) to reach the individual cross-referenced item directly. To return back to the point from where you accessed the cross-reference, press the ALT and keys
Trademarks
AC logo, Ardito, AudioCoded, AudioCodes, AudioCodes logo, CTI, CTI Squared, InTouch, IPmedia, Mediant, MediaPack, MP-MLQ, NetCoder, Netrake, Nuera, Open Solutions Network, OSN, Stretto, 3GX, TrunkPack, VoicePacketizer, VoIPerfect, What's Inside Matters, Your Gateway To VoIP, are trademarks or registered trademarks of AudioCodes Limited. All other products or trademarks are property of their respective owners.
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 support@audiocodes.com.
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 5.4
May 2008
MediaPack Series
Related Documentation
Document # Manual Name
LTRT-523xx (where xx denotes the Product Reference Manual document version) LTRT-654xx LTRT-598xx LTRT-665xx MP-11x and MP-124 SIP User's Manual MP-11x and MP-124 SIP-MGCP Installation Manual CPE Configuration Guide for IP Voice Mail
Notes: Throughout this manual, the terms MediaPack or device refer to the MP124, MP-118, MP-114, and MP-112 VoIP gateways. Throughout this manual, the term MP-11x refers to the MP-118, MP-114, and MP-112 MediaPack series VoIP gateways.
SIP Release Notes
SIP Release Notes
1. What's New in Release 5.4
What's New in Release 5.4

Note: This document uses a one-row table convention to indicate the products for which each feature is applicable. The products that don't support the feature are shaded (grayed). In the example below, the feature would be applicable only to MP-11x FXS.
MP-124
MP-11x
FXS
FXO
1.1
1.1.1
Supported Hardware Platforms

New Products Introduced in this Release
Not applicable.
1.1.2
Support of the Existing Hardware Platforms

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
1.1.3
Hardware Platforms No Longer Supported

Not applicable.
Version 5.4
May 2008
MediaPack Series
1.2
General Gateway New Features

The device supports the following new gateway features: 1. IPSec Tunneling Mode: MP-11x FXS FXO
MP-124
The device now supports IPSec Tunneling. This feature enables users to operate with IPSec when the conjunction endpoint device does not support IPSec. The secured channel is usually formed between AudioCodes' device and an IPSec-capable device (e.g., Firewall or VPN) in front of the endpoint. Relevant parameters: IPSecMode; IPSecPolicyRemoteTunnelIPAddress; IPsecPolicyRemoteSubnetMask. 2. Dead Peer Detection (DPD): MP-11x FXS FXO
MP-124
When two peers communicate using IPSec, connectivity between the two may go down unexpectedly. This can be due to, for example, routing problems or a rebooting of a host. In such cases, there is often no way for Internet Key Exchange (IKE) and IPSec to identify the loss of peer connectivity. However, the device can now detect loss of peer connectivity, by supporting the Dead Peer Detection (DPD) 'keepalive' mechanism according to RFC 3706. DPD uses IPSec traffic patterns to minimize the number of IKE messages required to confirm 'liveliness' of an IKE peer. Relevant parameter: IPSecDPDMode. 3. Additional Parameters in RADIUS Accounting Reports: MP-11x FXS FXO
MP-124
The following parameters were added to the RADIUS Accounting Report: Call-ID field which is identical to the Call-ID header used in the SIP INVITE message. This parameter was added to both the Setup and Release RADIUS reports to match the call with the accounting information. The call termination side. This parameter is needed to identify the side (IP or Tel) that terminated the call. This parameter was added to the Release RADIUS reports.
4.
Increased URL Length for CmpFileURL Parameter: MP-11x FXS FXO
MP-124
The maximum length of the URL address specified for Automatic Update Facility (i.e., automatic loading of a cmp file from a server) has been increased to 255 characters. Relevant parameter: CmpFileURL.
SIP Release Notes
SIP Release Notes
5.
Automatic Configuration URL Setup in Voice Menu: MP-11x FXS FXO
MP-124
The Voice Menu can now be used to configure an initial configuration URL on an HTTP server. 6. Enhanced G.711 (EG.711) Voice Coder: MP-11x FXS FXO
MP-124
The device now supports the Enhanced G.711 coder (EG.711). This coder is targeted for networks with high packet-loss ratio as it provides better voice quality. The highlights of this coder include the following: 7. Packet loss robustness. Variable frame length up to a maximum of 144 (10 msec) - 573 (40 msec) bytes. The frame length increases as the packet loss ratio increases. 8-kHz sampling rate. A-Law and -Law compatible.
Relevant parameters: CoderName; CoderName_ID. Special DTMF Representation: MP-11x FXS FXO
MP-124
The device now supports the ability to control the representation of special DTMF digits (* and #). These are used for out-of-band DTMF signaling (using SIP INFO/NOTIFY messages). These DTMFs can be depicted as characters or as numerical values (10 and 11). Relevant parameter: UseDigitForSpecialDTMF. 8. Mute RFC 2833 DTMF Digits after User-Defined Timeout: MP-11x FXS FXO
MP-124
It is now possible to define a maximum duration (timeout in milliseconds) for RFC 2833 Dual Tone Multi-frequency (DTMF/MF Relay) transmissions to the IP network, using the NTEMaxDuration parameter. If this timeout is reached, the RFC 2833 DTMF digital transmission is terminated (muted), regardless of the digit length received from the TDM side. Relevant parameter: NTEMaxDuration.
Version 5.4
May 2008
MediaPack Series 9. Additional CDR Fields: MP-11x FXS FXO
MP-124
The device now supports the following additional Call Detail Record (CDR) fields containing vital statistic information on calls made by the device: ReportType - report for either Call Started, Call Connected, or Call Released MeteringPulses - number of generated metering pulses RemotePackLoss - number of outgoing lost packets
10. Additional Parameters for IP Profile Feature: MP-124 MP-11x FXS FXO
The IP Profile provides additional parameters for controlling the following: Media Security Behavior (SRTP) - Preferable or Mandatory. Maximum number of concurrent calls: 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 for that specific profile (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 belonging to that profile. Copying Destination to Redirect Number.
Relevant parameter: IPProfile. 11. Enhanced Blind Transfer Keypad Feature: MP-124 MP-11x FXS FXO
The device supports a Blind Transfer keypad feature that allows a phone user to initiate a mid-call blind transfer by dialing a user-defined DTMF combination and then a transfer-to phone number. This feature has been modified to allow the user to press Hook-Flash before initiating a transfer. In addition, it is possible to add the Blind Transfer code as a prefix to the dialed destination number. Relevant parameters: KeyBlindTransfer; KeyBlindTransferAddPrefix. 12. Play of Secondary Dial Tone: MP-124 MP-11x FXS FXO
The device now supports the ability to play a secondary dial tone if a user-defined prefix is dialed from an FXS port. The device plays the regular dial tone and then starts collecting digits. If the External Line Prefix is identified (e.g., user dials 9 for an external line), a secondary dial tone is played and digit collection starts again. Relevant parameter: Prefix2ExtLine.
SIP Release Notes
10
SIP Release Notes
13. Reject Anonymous Calls: MP-124 MP-11x FXS FXO
The device now supports rejecting incoming anonymous calls. This capability can be enabled on a per FXS port basis or by using Keypad sequences for activation and deactivation. Relevant parameters: RejectAnonymousCallPerPort_ID; KeyRejectAnonymousCall; KeyRejectAnonymousCallDeact. 14. Emergency Call Support: MP-124 MP-11x FXS FXO
The device now supports the special handling of emergency calls (e.g., 911 and 112). A list of emergency numbers can be configured. A Tel-to-IP call is identified as an emergency call according to the destination number. The comparison is performed after number manipulation so that additional emergency numbers can be handled by the manipulation tables. SIP Priority and Resource-Priority headers are added to these emergency calls. If the user places the phone on-hook, the call is not disconnected but instead a Hold Re-INVITE request is sent to the remote party. The call is terminated only if the remote party disconnects the call (a BYE is received) or a user-defined timer expires. Relevant parameters: EmergencyNumbers; EmergencyRegretTimeout. 15. Stand Alone Survivability (SAS) Short Numbering: MP-124 MP-11x FXS FXO
The SAS Application has been enhanced to support Short Numbering. It may be convenient to dial an extension number only when calling within a local network / company. When SAS operates in emergency mode, it can perform suffix number matching. Users within the local network are registered to SAS with their global phone number / user info. Upon an incoming request, the user info in the request is compared to the registered one. Up to this release, only exact matching was acceptable. It is now possible to perform a comparison of the received user info with the suffix of the registered one. The length of the suffix is determined by the new parameter SASShortNumberLength. The registered (full) and dialed (full or short) numbers only match if one of the following conditions is met: The numbers fully match (i.e., same length and all characters match). The dialed number is shorter than the registered one, its length equals to the length specified by the parameter and it equals the suffix of the registered number determined by the length in the parameter.
Version 5.4
11
May 2008
MediaPack Series The table below provides a few examples of the SAS short numbering feature:
Parameter Value Not Relevant 0 4 3 or 5
Registered Number 12341234 12341234 12341234 12341234
Dialed Number 12341234 1234 1234 1234
Match? Yes No Yes No
Note: This feature operates in 'Emergency' mode only. Relevant parameter: SASShortNumberLength. 16. SAS TCP/TLS: MP-124 MP-11x FXS FXO
The SAS Application has been enhanced to support TCP and TLS as the transport layer (in addition to already supported UDP). Relevant parameters: SASLocalSIPTCPPort; SASLocalSIPTLSPort. 17. SIP OPTIONS for Verifying IP Connectivity: MP-124 MP-11x FXS FXO
The device now supports SIP OPTIONS requests to check connectivity with remote destinations. This is in addition to the existing capability of using ICMP Ping messages. Similar to Ping, OPTIONS requests are sent periodically to user-defined IP destinations, and the connectivity status is derived from the responses. The entity is considered Offline if the last completed OPTIONS transaction has timed out. Any response to an OPTIONS request, even if indicating an error, returns the connectivity status to Online. Relevant parameter: AltRoutingTel2IPConnMethod. 18. Activation of Busy Out when No Connectivity to All Destinations: MP-124 MP-11x FXS FXO
When both the Busy Out and the IP Connectivity mechanisms are enabled and there is no connectivity to any destination IP address, the device is placed in Busy Out mode. Relevant parameter: EnableBusyOut.
SIP Release Notes
12
SIP Release Notes
19. Configurable Transport Type for Servers and IP Addresses: MP-124 MP-11x FXS FXO
The device now supports specifying the requested Transport Type (UDP, TCP, or TLS) for the following servers and IP addresses: Proxy servers Destination IP address in the 'Tel-to-IP Routing' table Registrar MWI server
Note: If a Transport Type is defined, no NAPTR query is performed (if the device is configured to use NAPTR). Relevant parameters: ProxySet; Prefix; RegistrarTransportType; MWIServerTransportType; . 20. Unencrypted Secure RTP Control Protocol (SRTCP): MP-124 MP-11x FXS FXO
The device now supports the option to declare that SRTCP not be encrypted when configured to use SRTP/SRTCP. Note: When configured, the device adds the Unencrypted_SRTCP attribute to the crypto attribute of the outgoing SDP (SRTCP is still encrypted). To disable the sending of RTCP packets, set the parameter RTCPInterval to 0. Relevant parameter: OfferUnencryptedSRTCP. 21. Multi-Level Precedence & Preemption (MLPP) Support on Analog Devices: MP-124 MP-11x FXS FXO
The device now supports the MLPP protocol for analog endpoints. Using this protocol, (1) each call can be assigned a precedence value; (2) calls can be preempted by higher precedence calls. For a full description of the MLPP implementation using SIP, refer to the device's User's Manual. Relevant parameters: FirstCallRBTId; PrecedenceRingingType. 22. Generate Wink Upon Receipt of Hook-Flash INFO: MP-124 MP-11x FXS FXO
The device now supports generating a Wink signal to the FXS port upon receiving a SIP INFO message containing a Hook-Flash signal. The Wink duration is configured using the parameter FlashHookPeriod. Relevant parameter: FlashHookPeriod.
Version 5.4
13
May 2008
MediaPack Series
1.3
SIP New Features

The device supports the following new SIP features: 1. Retry-After SIP Header: MP-11x FXS FXO
MP-124
The device now supports adding and responding to the SIP Retry-After header (defined in RFC 3261). When acting as a UAC and receiving a 503 (Service Unavailable) response with a Retry-After header, no new dialogs are established (including INVITE, REGISTER, and OPTIONS requests) towards the UAS until the time defined in the Retry-After header expires. When acting as a UAS, in scenarios where the device generates a 503 response (Overload state), a Retry-After header is added (if configured). Relevant parameter: RetryAfterTime. 2. Restrict Calling Party Number: MP-11x FXS FXO
MP-124
The device now supports setting to anonymous the User-Part of the Contact header for outgoing INVITE requests on restricted calls. Relevant parameter: EnableContactRestriction. 3. RFC 3455 (Private Header Extensions to SIP for the 3GPP) [Partial]: MP-11x FXS FXO
MP-124
The device now supports receiving and usage of the P-Associated-URI header. The P-Associated-URI header allows a registrar to return a set of associated URIs for a registered Address-of-Record. This header is used in the 200 OK response to a REGISTER request. Handling of the P-Associated-URI header is determined by a new configuration parameter (EnablePAssociatedURIHeader). P-Associated-URIs in registration responses are handled only if the device is registered per endpoint. In case of registration per device (gateway), the P-Associated-URI header is not handled and the From header value is not changed in future requests. Only the first URI in the P-Associated-URI header is used in subsequent requests. In case of replacing the From header value with the P-Associated-URI value, the following requests and responses are affected: INVITE: From header, P-Asserted-Id Header. SUBSCRIBE: From header, P-Asserted-Id Header. 200 OK: P-Asserted Header.
Relevant parameter: EnablePAssociatedURIHeader.
SIP Release Notes
14
SIP Release Notes
4.
RFC 4235 (An INVITE-Initiated Dialog Event Package for SIP) [Partial]: MP-11x FXS FXO
MP-124
The device now partially supports the SIP Dialog package defined in RFC 4235. The dialog package allows users to subscribe to another user in order to receive notifications of the changes in the state of INVITE-initiated dialog usages in which the subscribed-to user is involved. The device supports receiving subscriptions per endpoint only. The device generates NOTIFY messages according to changes in the endpoint state: 5. Idle - no dialogs on the endpoint (no <state> element). Seized/Answered - either handset is off-hook or the endpoint is in the middle of a confirmed session (<state>confirmed</state>).
RFC 3680 (A SIP Event Package for Registrations): MP-11x FXS FXO
MP-124
The device now supports the SIP Event package for registrations as defined in RFC 3680. This definition allows a SIP endpoint to receive notifications of its registration state by the Registrar. The device does not send SUBSCRIBE requests, but handles unsolicited NOTIFY messages that are received from the Registrar. The device supports these notifications when operating in registration per endpoint or per device. The RFC defines several different states, each of which is handled differently by the device: 6. Created, Registered, Refreshed, Unregistered: ignored by the device. Deactivated, Shortened, Expired: a new REGISTER request is sent to the Registrar using the same Call-Id as the previous request. Rejected: the device stops sending registration request to the server and switches to the next server on the Proxy list. Probation: the device resumes registrations after the Retry-After time expires.
Set OPTIONS Request-URI User-Part: MP-11x FXS FXO
MP-124
It is now possible to define the Request-URI User-Part value, which is used in outgoing SIP OPTIONS requests. Relevant parameter: OPTIONSUserPart.
Version 5.4
15
May 2008
MediaPack Series 7. TCP and TLS Enhancements: MP-11x FXS FXO
MP-124
The following enhancements were added to the device's TCP and TLS capabilities: SIP connection persistency is now performed for TCP connections (as well as TLS) when the connection is initiated to or from the Proxy. The device now supports re-Registration after renewal of a TCP or TLS connection. The device now supports TLS Re-negotiation using a configurable interval. TLS Server Certificate verification can be performed regardless if Mutual Authentication is required. Subject Name verification is now supported when establishing TLS connections using AltSubjectName and CN fields.
Relevant parameters: ReRegisterOnConnectionFailure; TLSReHandshakeInterval; SIPSRequireClientCertificate; PeerHostNameVerificationMode; VerifyServerCertificate; TLSRemoteSubjectName. 8. Enhanced Call Routing Configuration Options: MP-11x FXS FXO
MP-124
The call routing configuration has been enhanced to provide the user with more flexibility and greater control over Tel-to-IP and IP-to-Tel call routing. The enhanced configuration allows users to define various entities (IP Groups and Proxy Sets). The Proxy Sets include groups of user-defined Proxy servers (IP address or FQDN) and the IP Groups are logical entities to which these Proxy Sets can be assigned. These IP Groups can then later be assigned to Hunt Groups in various tables (i.e., Tel-to-IP Routing, IP-to-Trunk Group Routing, Trunk Group Settings, and Accounts tables). The new Accounts table implements these entities to enable flexible registration and authentication configuration, allowing registration and authentication per device's Hunt Groups (i.e., per "account"). Relevant parameters: Account; Prefix; PSTNPrefix; TrunkGroupSettings; IPGroup; ProxyIP.
SIP Release Notes
16
SIP Release Notes
1.4
Web and SNMP New Features

The device supports the following new Web and SNMP features: 1. Web Interface Scenario Feature: MP-11x FXS FXO
MP-124
The Web interface now offers a Scenario feature. This feature allows the user to create a customized 'menu' (referred to as a Scenario) that can include up to 20 configuration pages (selected from the Web interface menus). For each page in the Scenario (referred to as a Step), the user can select up to 25 parameters pertaining to the configuration page. Therefore, the Scenario feature is useful in that it provides the user with quick-and-easy access to commonly used configuration parameters specific to the user's network environment The Web interface also allows the user to edit and delete a Scenario, as well as save and load a Scenario to and from a PC respectively. Relevant parameter: ScenarioFileName. 2. Web Interface Online Help: MP-11x FXS FXO
MP-124
The Web interface now provides a context-sensitive Online Help. The Online Help can be accessed by clicking the Help icon located on the toolbar. The Online Help displays brief descriptions of topics and parameters pertaining to the currently opened Web page. 3. Public Key Infrastructure Certificate: MP-11x FXS FXO
MP-124
The following SNMP parameters were added: acSysActionSetIPSecTLSUpgrade: updates the IPSec TLS configuration according to the provisioned TLS URL file acSysActionSetGWAppTLSUpgrade: updates the GWApp TLS configuration according to the provisioned TLS URL file
This was done using acSysHTTPClientTLSPkeyFileUrl, acSysHTTPClientTLSCertFileUrl, and acSysHTTPClientTLSRootFileUrl which were previously added. 4. Cold Start Indicator - New MIB Leaf: MP-11x FXS FXO
MP-124
The new MIB object acSysSNMPEmsColdStartIndication has been added. If a device reset occurs, this MIB value is set to zero. This parameter has no affect on the device. If there is a need for the EMS to use this parameter, its value must be set using the EMS to a value other than its default value of 0.
Version 5.4
17
May 2008
MediaPack Series
1.5
New Parameters
The table below describes the new parameters for Release 5.4. Most of these new parameters can be configured using both the ini file (enclosed in square brackets) and the Web interface. Table 1-1: Release 5.4 New Web / [ini] File Parameters
Web / [ini] File Parameter Name IPSec Mode [IPSecMode]
Description Defines the IPSec mode of operation.

[0] Transport (Default) [1] Tunneling
Remote Tunnel IP Defines the IP address of the remote IPSec tunneling device. Address Note: This parameter is only available if the parameter IPSecMode is set [IPSecPolicyRemoteTun to Tunneling (1). nelIPAddress] Remote Subnet Mask Defines the subnet mask of the remote IPSec tunneling device. [IPsecPolicyRemoteSub The default value is 255.255.255.255 (i.e., host-to-host IPSec tunnel). netMask] Note: This parameter is only available if the parameter IPSecMode is set to Tunneling (1). Dead Peer Detection Mode [IPSecDPDMode] Enables the Dead Peer Detection (DPD) 'keepalive' mechanism (according to RFC 3706) to detect loss of IKE peer connectivity.

[0] Disabled (default). [1] Periodic = message exchanges at regular intervals. [2] On Demand = message exchanges as needed (i.e., before sending data to the peer). If the liveliness of the peer is questionable, the device sends a DPD message to query the status of the peer. If the device has no traffic to send, it never sends a DPD message.
Special Digit Defines the representation for special digits (* and #) that are used for Representation out-of-band DTMF signaling (using SIP INFO/NOTIFY). [UseDigitForSpecialDTM [0] Special = Uses the strings * and # (default). F] [1] Numeric = Uses the numerical values 10 and 11. [NTEMaxDuration] Maximum time for sending Named Telephony Events (NTEs) to the IP side, regardless of the time range when the TDM signal is detected. The range is -1 to 200,000,000 msec (i.e., 55 hours). The default is -1 (i.e., NTE stops only upon detection of an End event).
[KeyBlindTransferAddPr Determines whether the device adds the Blind Transfer code (KeyBlindTransfer) to the dialed destination number. efix]

[0] Disable (default). [1] Enable.
Note: This parameter is applicable only to FXS interfaces. [Prefix2ExtLine] Defines a string prefix (e.g., '9') that when dialed from an FXS port causes the device's FXS port to play a secondary dial tone and then restart digit collection. The valid range is a 1-character string. The default is an empty string. Note: This parameter is applicable only to FXS interfaces.
SIP Release Notes
18
SIP Release Notes
Web / [ini] File Parameter Name
Description
[RejectAnonymousCallP Determines whether the device rejects incoming anonymous calls on FXS interfaces. erPort_ID] The format for this ini file table parameter is as follows: [RejectAnonymousCallPerPort] FORMAT RejectAnonymousCallPerPort_Index = RejectAnonymousCallPerPort_Enable; [\RejectAnonymousCallPerPort] Where,
Enable = accept [0] (default) or reject [1] incoming anonymous calls.
For example: [RejectAnonymousCallPerPort] RejectAnonymousCallPerPort 0 = 0; RejectAnonymousCallPerPort 1 = 1; [\RejectAnonymousCallPerPort] If enabled, when a device's FXS interface receives an anonymous call, it responds with a 433 (Anonymity Disallowed) SIP response. Notes:

This parameter is applicable only to FXS interfaces. This parameter is per device. This parameter can appear up to eight times for 8-port MP-11x devices and up to 24 times for MP-124 devices. The double dollar ($$) symbol represents the default value.
Activate Keypad sequence that activates the reject anonymous call option, [KeyRejectAnonymousC whereby the device rejects incoming anonymous calls. After the sequence is pressed, a confirmation tone is heard. all] Note: This parameter is applicable only to FXS interfaces. Deactivate Keypad sequence that de-activates the reject anonymous call option. After [KeyRejectAnonymousC the sequence is pressed, a confirmation tone is heard. allDeact] Note: This parameter is applicable only to FXS interfaces. Emergency Numbers [EmergencyNumbers] Defines a list of numbers which are defined as 'emergency' numbers. When one of these numbers is dialed, the outgoing INVITE message includes the Priority and Resource-Priority headers. If the user sets the phone on-hook, the call is not disconnected, but instead a Hold ReINVITE request is sent to the remote party. Only if the remote party disconnects the call (i.e., a BYE is received) or a timer expires (set by the parameter EmergencyRegretTimeout) is the call terminated. The list can include up to four different numbers, where each number is up to four digits long. For example: EmergencyNumbers = 100,911,112 Note: This parameter is applicable only to FXS interfaces. Emergency Calls Reanswer Timeout [EmergencyRegretTime out] Determines the time (in minutes) that the device waits before tearingdown an emergency call (defined by the parameter EmergencyNumbers). Until this time expires, an emergency call can only be disconnected by the remote party [(typically, by a Public Safety Answering Point (PSAP)]. The valid range is 1 to 30. The default value is 10. Note: This parameter is applicable only to FXS interfaces.
Version 5.4
19
May 2008
MediaPack Series Web / [ini] File Parameter Name
Description
Short Number Length Determines the length of the suffix used for Stand-Alone Survivability [SASShortNumberLengt (SAS) Short Numbering support. The dialed number is compared to the registered number. The registered (full) and dialed (full or short) numbers h] match only if one of the following conditions is met:

The numbers match entirely (i.e., they are of the same length and all characters match). The dialed number is shorter than the registered one, its length equals to the length specified by the parameter and it equals the suffix of the registered number determined by the length in the parameter.
When set to 0, only a full match between the dialed and registered numbers is valid. The valid range is 0 to 63 characters. The default value is no character. Note: This parameter is applicable only in the SAS Emergency mode SAS Local SIP TCP Port [SASLocalSIPTCPPort] Local TCP port used to send/receive SIP messages for the SAS application. The SIP entities in the local network need to send the registration requests to this port. When forwarding the requests to the proxy ('Normal Mode'), this port serves as the source port. The valid range is 1 to 65,534. The default value is 5,080. Note: This parameter is applicable only in the SAS Emergency mode SAS Local SIP TLS Port [SASLocalSIPTLSPort] Local TLS port used to send/receive SIP messages for the SAS application. The SIP entities in the local network need to send the registration requests to this port. When forwarding the requests to the proxy ('Normal Mode'), this port serves as the source port. The valid range is 1 to 65,534. The default value is 5,081. Note: This parameter is applicable only in the SAS Emergency mode Alt Routing Tel to IP Determines the method used by the device for periodically querying the Connectivity Method connectivity status of a destination IP address. [AltRoutingTel2IPConnM [0] ICMP Ping (default) = Internet Control Message Protocol (ICMP) ethod] ping messages.
[1] SIP OPTIONS = The remote destination is considered offline if the latest OPTIONS transaction timed out. Any response to an OPTIONS request, even if indicating an error, brings the connectivity status to online.
MWI Server Transport Determines the transport layer used for outgoing SIP dialogs initiated by Type the device to the MWI Server. [MWIServerTransportTy [-1] Not Configured (default) pe] [0] UDP

[1] TCP [2] TLS
Note: When set to Not Configured (-1), the value of the parameter SIPTransportType is used.
SIP Release Notes
20
SIP Release Notes
Description
Registrar Transport Type Determines the transport layer used for outgoing SIP dialogs initiated by [RegistrarTransportType the device to the Registrar. ] [-1] Not Configured (default)

[0] UDP [1] TCP [2] TLS
Note: When set to Not Configured, the value of the parameter SIPTransportType is used. [OfferUnencryptedSRTC Determines whether the device adds the UNENCRYPTED_SRTCP attribute to outgoing SDP messages when SRTP/SRTCP is enabled. P]

For example: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WxMz3YE0fcbjVJOGgSxweCZxySisl6SWi103t6No UNENCRYPTED_SRTCP Note: When enabled, the device adds the UNENCRYPTED_SRTCP attribute to the crypto attribute of the outgoing SDP (SRTCP is still encrypted). To disable the sending of RTCP packets, set the parameter RTCPInterval to 0. [FirstCallRBTId] Defines the index of the first Ringback Tone in the Call Progress Tones (CPT) file. The valid range is -1 to 1,000. The default value is -1 (play standard Ringback tone).
[PrecedenceRingingTyp Defines the index of the Precedence Ringing tone in the Call Progress Tones (CPT) file. This tone is used when CallPriorityMode is set to 1 and e] a Precedence call is received from the IP side. The valid range is -1 to 16. The default value is -1 (plays standard Ringing tone). Retry-After Time [RetryAfterTime] Determines the time (in seconds) used in the Retry-After header when a 503 (Service Unavailable) is generated by the device. The time range is 0 to 3,600. The default value is 0.
Enable Contact Determines whether or not the device sets the Contact header of outgoing Restriction INVITE requests to anonymous for restricted calls. [EnableContactRestricti [0] = Disabled (default) on] [1] = Enabled Enable P-Associated-URI Header [EnablePAssociatedURI Header] Determines the device usage of the P-Associated-URI header. This header can be received in 200 OK responses to REGISTER requests. When enabled, the first URI in the P-Associated-URI header is used in subsequent requests as the From / P-Asserted-Id headers value.

Note: P-Associated-URIs in registration responses is handled only if the device is registered per endpoint.
Version 5.4
21
May 2008
MediaPack Series Web / [ini] File Parameter Name [OPTIONSUserPart]
Description Defines the User-Part value of the Request-URI for outgoing SIP OPTIONS requests. If no value is configured, the endpoint number is used. A special value is empty, indicating that no User-Part in the Request-URI (Host-Part only) is used. The valid range is a 30-character string. The default value is an empty string ().
[ReRegisterOnConnecti Enables the device to perform SIP Re-Registration upon TCP/TLS connection failure. onFailure]

[TLSReHandshakeInterv Defines the time interval (in minutes) between TLS Re-Handshake operations initiated by the device. al] The interval range is 0 to 1,500 minutes. The default is 0 (i.e., no TLS ReHandshake). [PeerHostNameVerificati Determines whether the device verifies the Subject Name of a remote certificate when establishing TLS connections. onMode]

[0] = Disable (default). [1] = Verify Subject Name only when acting as a server for the TLS connection. [2] = Verify Subject Name when acting as a server or client for the TLS connection.
When a remote certificate is received and this parameter is not disabled, the SubjectAltName value is compared with the list of available Proxies. If a match is found for any of the configured Proxies, the TLS connection is established. The comparison is performed if the SubjectAltName is either a DNS name (DNSName) or an IP address. If no match is found and the SubjectAltName is marked as critical, the TLS connection is not established. If the SubjectAltName is not marked as critical and there is no match, the CN value of the SubjectName field is compared with the parameter TLSRemoteSubjectName. If a match is found, the connection is established. Otherwise, the connection is terminated. [VerifyServerCertificate] Determines whether the device, when acting as client for TLS connections, verifies the Server certificate. The certificate is verified with the Root CA information.

Note: If Subject Name verification is necessary, the parameter PeerHostNameVerificationMode must be used as well.
SIP Release Notes
22
SIP Release Notes
Description
[TLSRemoteSubjectNam Defines the Subject Name that is compared with the name defined in the remote side certificate when establishing TLS connections. e] If the SubjectAltName of the received certificate is not equal to any of the defined Proxies Host names/IP addresses and is not marked as 'critical', the Common Name (CN) of the Subject field is compared with this value. If not equal, the TLS connection is not established. The valid range is a string of up to 49 characters. Note: This parameter is applicable only if the parameter PeerHostNameVerificationMode is set to 1 or 2. [IPGroup] This ini file table parameter configures the IP Group table. The format of this parameter is as follows: [IPGroup] FORMAT IPGroup_Index = IPGroup_Type, IPGroup_Description, IPGroup_ProxySetId, IPGroup_SIPGroupName, IPGroup_ContactUser, IPGroup_EnableSurvivability, IPGroup_ServingIPGroup, IPGroup_SendInviteToProxy, IPGroup_AlwaysUseRouteTable; [\IPGroup] Where,

Description = Brief description of the IP Group. ProxySetId = Proxy Set ID associated with the IP Group. SIPGroupName = Request URI host name used in INVITE and REGISTER messages. SendInviteToProxy = If enabled (1), the INVITE as a result of REFER/3xx messages are sent to the IP Group, overriding the ReferTo/Contact destination. AlwaysUseRouteTable = If enabled (1), the device uses the IP address (or domain name) defined in the 'Tel to IP Routing' table as the Request URI host name in outgoing INVITE messages, instead of the value entered in the SIPGroupName.
For example: [IPGroup] FORMAT IPGroup_Index = IPGroup_Type, IPGroup_Description, IPGroup_ProxySetId, IPGroup_SIPGroupName, IPGroup_ContactUser, IPGroup_EnableSurvivability, IPGroup_ServingIPGroup, IPGroup_SendInviteToProxy, IPGroup_AlwaysUseRouteTable; IPGroup 1 = 0, "Verizon gateway", 1, firstIPgroup, , 0, -1, 0, 0; IPGroup 2 = 0, "Avaya server", 2, secondIPgroup, , 0, -1, 0, 0; IPGroup 3 = 0, "IP phones", 1, thirdIPGroup, , 0, -1, 0, 0; [\IPGroup] Notes:

This table parameter can include up to 9 indices (1-9). The parameters IPGroup_Type, IPGroup_ContactUser, IPGroup_EnableSurvivability, and IPGroup_ServingIPGroup are currently not applicable and must be left empty (or -1). These parameters are used only for IP-to-IP call routing applications (supported in the next applicable release).
Version 5.4
23
May 2008
MediaPack Series Web / [ini] File Parameter Name [Account]
Description This ini file table parameter configures the Account table for registering and/or authenticating (digest) a Trunk Group (e.g., IP-PBX) to a Serving IP Group (e.g., Internet Telephony Service Provider - ITSP). The format of this parameter is as follows: [Account] FORMAT Account_Index = Account_ServedTrunkGroup, Account_ServedIPGroup, Account_ServingIPGroup, Account_Username, Account_Password, Account_HostName, Account_Register, Account_ContactUser; [\Account] Where,

ServedTrunkGroup = Trunk Group ID for which the device performs registration/authentication to a destination IP Group. ServedIPGroup = Currently not applicable (see note below). ServingIPGroup = Destination IP (Group) to where the device sends the REGISTER requests (and/or digest authentication username and password) for registering (and/or authenticating) the Trunk Group. Username = Digest authentication user name. Password = Digest authentication password. HostName = Register request URI host name sent by the device to the Serving IP Group. Register = Enables registration mode (i.e., device sends REGISTER requests to Serving IP Group). ContactUser = AOR user name.

For example: [Account] FORMAT Account_Index = Account_ServedTrunkGroup, Account_ServedIPGroup, Account_ServingIPGroup, Account_Username, Account_Password, Account_HostName, Account_Register, Account_ContactUser; Account 0 = 1, -1, 1, user, 1234, acl, 1, ITSP1; [\Account] Notes:

This table can include up to 10 indices. The table item Account_ServedIPGroup is currently not applicable and must be left empty (or assigned the value -1). It is used only for IP-toIP routing applications (supported in the next applicable release). You can define multiple table indices having the same ServedTrunkGroup with different ServingIPGroups, Username, Password, HostName, and ContactUser. This provides the capability for registering the same Trunk Group to several ITSP's (i.e., Serving IP Groups).
[ScenarioFileName]
Defines the file name of the Scenario to be loaded to the device. The file name must have the *.dat file extension and can be up to 47 characters.
SIP Release Notes
24
SIP Release Notes
Web / [ini] File Parameter Name [ZeroSDPHandling]
Description Determines the device's response to an incoming SDP with an IP address of 0.0.0.0 in the Connection line.

[0] Sets the IP address of the outgoing SDP Connection line to 0.0.0.0 (default). [1] Sets the IP address of the outgoing SDP Connection line to the device's own IP address and adds a 'a=sendonly' line to the SDP.
Digit To Ignore Digit A digit pattern that if received as Src (S) or Redirect (R) numbers is Pattern ignored and not added to that number. [DigitPatternDigitToIgno The valid range is a 25-character string. re] [HandleReasonHeader] Determines whether the device uses the value of the incoming SIP Reason header for Release Reason mapping.

[0] Disregard Reason header in incoming SIP messages. [1] Use the Reason header value for Release Reason mapping (default).
[EnableSecureStartup]
Enables the Secure Startup mode. In this mode, downloading the *.ini file to the device is restricted to a URL provided in initial configuration (see parameter IniFileURL) or using DHCP.

[0] Disable (default). [1] Enable = disables TFTP and allows secure protocols such as HTTPS to fetch the device configuration.
Note: For a detailed explanation on Secure Startup, refer to the Product Reference Manual. [FaxCNGMode] Determines the device's behavior upon detection of a CNG tone.

[0] = Does not send a SIP Re-INVITE upon detection of a fax CNG tone when CNGDetectorMode is set to 1 (default). [1] = Sends a SIP Re-INVITE upon detection of a fax CNG tone when CNGDetectorMode is set to 1. [0] = Send all Syslog messages to the defined Syslog server (default). [1] = Send all Syslog messages using the Debug Recording mechanism. [2] = Send only Error and Warning level Syslog messages using the Debug Recording mechanism.
[SyslogOutputMethod]
Determines the method used for Syslog messages.

[SourceNumberPreferen Determines the SIP header used to determine the Source Number in incoming INVITE messages. ce]

(empty string) = Use device's internal logic for header preference (default). FROM = Use the Source Number received in the From header.
The valid range is a string of up to 10 characters. The default is an empty string.
Version 5.4
25
May 2008
Description
[EnableSilenceSuppInS Determines the device's behavior upon receipt of SIP Re-INVITE messages that include the silencesupp:off attribute. DP]

[0] = Disregard the silecesupp attribute (default). [1] = Handle incoming Re-INVITE messages that include the silencesupp:off attribute in the SDP as a request to switch to the VoiceBand-Data (VBD) mode.
[ForkingHandlingMode] Determines how the device reacts to forking of outgoing INVITE messages by the Proxy.
[0] = Sequential. The device opens a voice stream toward the first 18x SIP response that includes an SDP, and disregards any 18x response with an SDP received thereafter (default). [1] = Parallel. The device opens a voice stream toward the first 18x SIP response that includes an SDP, and re-opens the stream toward any subsequent 18x responses with an SDP.
Note: Regardless of the ForkingHandlingMode value, once a 200 OK response is received, the device uses the RTP information and re-opens the voice stream, if necessary. [SourceIPAddressInput] Determines the IP address which the device uses to decide the source of incoming INVITE messages for routing.

[-1] = Not configured (default). [0] = Use the IP address received in the Contact header of the incoming INVITE message. [1] = Use the actual IP address (Layer 3) from which the SIP packet was received.
[BlindTransferDisconne Defines the duration (in milliseconds) for which the device waits for a disconnection from the Tel side after the Blind Transfer Code ctTimeout] (KeyBlindTransfer) has been identified. When this timer expires, a SIP REFER message is sent toward the IP side. If this parameter is set to 0, the REFER message is immediately sent. The valid range is 0 to 1,000,000. The default is 0. [MLPPDiffserv] Defines the DiffServ value (DSCP) used in IP packets containing SIP messages that are related to MLPP calls. The valid range is 0 to 63. The default value is 50. Enables Priority Calls handling.

[CallPriorityMode]
[0] Disable (default). [1] MLPP = Priority Calls handling is enabled.
SIP Release Notes
26
SIP Release Notes
Web / [ini] File Parameter Name ProxySet
Description This ini file table parameter configures the Proxy Set table by assigning various attributes per Proxy Set ID. The format of this parameter is as follows: [ProxySet] FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive, ProxySet_ProxyKeepAliveTime, ProxySet_ProxyLoadBalancingMethod, ProxySet_IsProxyHotSwap; [\ProxySet] For example: [ProxySet] FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive, ProxySet_ProxyKeepAliveTime, ProxySet_ProxyLoadBalancingMethod, ProxySet_IsProxyHotSwap; ProxySet 0 = 0, 60, 0, 0; ProxySet 1 = 1, 60, 1, 0; [\ProxySet] Notes:

This table parameter can include up to 6 indices (0-5). For configuring the Proxy Sets, refer to the ini file parameter ProxyIP.
Version 5.4
27
May 2008
MediaPack Series
1.6
Modified Parameters
The table below lists parameters from the previous release that have been modified for Release 5.4. The parameters enclosed in square brackets depict the ini file parameter; the other parameters depict the parameters in the Embedded Web Server. Table 1-2: Release 5.4 Modified Web / [ini] File Parameters
Web / [ini] File Parameter Name [CmpFileURL]
Description (Modification: Maximum URL address length increased.) Specifies the name of the cmp file and the location of the server (IP address or FQDN) from which the device loads a new cmp file and updates itself. The cmp file can be loaded using HTTP, HTTPS, FTP, FTPS, or NFS. For example: http://192.168.0.1/filename Notes:

When this parameter is set in the ini file, the device always loads the cmp file after it is reset. The cmp file is validated before it's burned to flash. The checksum of the cmp file is also compared to the previously-burnt checksum to avoid unnecessary resets. The maximum length of the URL address is 255 characters.
[IPProfile]
(Modification: Addition of parameters MediaSecurityBehaviour and CallLimit.) This ini file parameter table configures the IP profiles table and has the following format: [IPProfile] FORMAT IPProfile_Index = IPProfile_ProfileName, IPProfile_IpPreference, IPProfile_CodersGroupID, IPProfile_IsFaxUsed*, IPProfile_JitterBufMinDelay*, IPProfile_JitterBufOptFactor*, IPProfile_IPDiffServ*, IPProfile_SigIPDiffServ*, N/A, IPProfile_RTPRedundancyDepth, IPProfile_RemoteBaseUDPPort, IPProfile_CNGmode, IPProfile_VxxTransportType, IPProfile_NSEMode, N/A, IPProfile_PlayRBTone2IP, IPProfile_EnableEarlyMedia*, IPProfile_ProgressIndicator2IP*, IPProfile_EnableEchoCanceller*, IPProfile_MediaSecurityBehaviour, IPProfile_CallLimit; [\IPProfile] For example: [IPProfile] IPProfile_1 = name1,2,1,0,10,13,15,44,1,1,6000,0,2,0,0,0,1,0,1,,0,-1; IPProfile_2 = name2,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,$$,40; [\IPProfile] Notes:

* Indicates common parameters used in both IP and Tel profiles. The parameter IpPreference determines the priority of the Profile (1 to 20, where 20 is the highest preference). If both IP and Tel profiles apply to the same call, the coders and other common parameters (indicated with an asterisk) of the preferred Profile are applied to that call. If the Tel and IP profiles are identical, the Tel Profile parameters are applied. Two adjacent dollar signs ('$$') indicate that the parameter's default 28 Document #: LTRT-65610
SIP Release Notes
SIP Release Notes
Web / [ini] File Parameter Name value is used.

Description
IPProfile can be used in the 'Tel to IP Routing' and 'IP to Hunt Group Routing' tables (Prefix and PSTNPrefix parameters). The 'Profile Name' assigned to a Profile index, must enable users to identify it intuitively and easily. This parameter can appear up to 9 times (i.e., indices 1 through 9).
Blind [KeyBlindTransfer]
(Modification: Description of transfer process and addition of a note.) Keypad sequence that activates the blind transfer option. There are two possible scenarios:
Option 1: After this sequence is dialed, the current call is put on hold, a dial tone is played to the phone, and then phone number collection starts. Option 2: A Hook-Flash is pressed, the current call is put on hold, a dial tone is played to the phone, and then digit collection starts. After this sequence is identified, the device continues the collection of the destination phone number.
For both options, after the phone number is collected, it's sent to the transferee in a SIP REFER request (without a Replaces header). The call is then terminated and a confirmation tone is played to the phone. If the phone number collection fails due to a mismatch, a reorder tone is played to the phone. Notes:

This parameter is applicable only to FXS interfaces. You can configure whether the KeyBlindTransfer code is added as a prefix to the dialed destination number, by using the parameter KeyBlindTransferAddPrefix.
Version 5.4
29
May 2008
MediaPack Series Web / [ini] File Parameter Name Enable Busy Out [EnableBusyOut]
Description (Modification: Addition of the no connectivity to any destination option.) Determines whether the Busy Out feature is enabled.

[0] Disable = 'Busy out' feature is not used (default). [1] Enable = 'Busy out' feature is enabled.
When Busy Out is enabled and certain scenarios exist, the device performs the following: A reorder tone (determined by FXSOOSBehavior) is played when the phone is off-hooked. These behaviors are performed due to one of the following scenarios: Physically disconnected from the network (i.e., Ethernet cable is disconnected).
The Ethernet cable is connected, but the device can't communicate with any host. Note that LAN Watchdog must be activated (EnableLANWatchDog = 1). The device can't communicate with the proxy (according to the Proxy keep-alive mechanism) and no other alternative exists to send the call. The IP Connectivity mechanism is enabled and there is no connectivity to any destination IP address. The FXSOOSBehavior parameter controls the behavior of the FXS endpoints when a Busy Out or Graceful Lock occurs. FXO endpoints during Busy Out and Lock are inactive. Refer to the LifeLineType parameter for complementary optional behavior.

Notes:

[Prefix]
(Modification: New parameters Prefix_DestIPGroupID and Prefix_TransportType) Configures the 'Tel to IP Routing' table to route incoming Tel calls to IP addresses. The format of this ini file table parameter is as follows: [PREFIX] FORMAT PREFIX_Index = PREFIX_DestinationPrefix, PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId, PREFIX_MeteringCode, PREFIX_SrcIPGroupID, PREFIX_DestHostPrefix, PREFIX_DestIPGroupID, PREFIX_SrcHostPrefix, PREFIX_TransportType, PREFIX_SrcTrunkGroupID; [\PREFIX] Where,

DestinationPrefix = Destination phone prefix. DestAddress = Destination IP address. SourcePrefix =Source phone prefix. ProfileID = Profile ID number. MeteringCode = Charge code. DestHostPrefix = N/A. DestIPGroupID = IP Group (1-9) to where you want to route the Tel-to-IP call. SrcHostPrefix = N/A. 30 Document #: LTRT-65610
SIP Release Notes
SIP Release Notes

Description TransportType = Destination transport type: Not configured (-1), the global SIPTransportType type is used; UDP (0); TCP (1); TLS (2). SrcTrunkGroupID = N/A.
For example: [PREFIX] FORMAT PREFIX_Index = PREFIX_DestinationPrefix, PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId, PREFIX_MeteringCode, PREFIX_SrcIPGroupID, PREFIX_DestHostPrefix, PREFIX_DestIPGroupID, PREFIX_SrcHostPrefix, PREFIX_TransportType, PREFIX_SrcTrunkGroupID; PREFIX 0 = *, quest, *, 0, 255, -1, , 1, , -1, -1; PREFIX 1 = 20, 10.33.37.77, *, 0, 255, -1, , 2, , 0, -1; PREFIX 2 = 30, 10.33.37.79, *, 1, 255, -1, , -1, , 2, -1; [\PREFIX] Notes:

This parameter can include up to indices. The parameters SrcIPGroupID, DestHostPrefix, and SrcHostPrefix are currently not applicable and must be left empty (or -1). (They are used only for IP-to-IP routing, supported in the next applicable release). The phone prefix for destination (DestinationPrefix) and source (SourcePrefix) addresses can be a single number or a range of numbers. Parameters can be skipped using two dollar ($$) symbols, for example: Prefix = $$,10.2.10.2,202,1. The IP address can include wildcards. The 'x' wildcard is used to represent single digits, e.g., 10.8.8.xx represents all addresses between 10.8.8.10 to 10.8.8.99. The '*' wildcard represents any number between 0 and 255, e.g., 10.8.8.* represents all addresses between 10.8.8.0 and 10.8.8.255. If the string 'ENUM' is specified for the destination IP address, an ENUM query containing the destination phone number is sent to the DNS server. The ENUM reply includes a SIP URI used as the Request-URI in the outgoing INVITE and for routing (if Proxy is not used).

Version 5.4
31
May 2008
MediaPack Series Web / [ini] File Parameter Name [PSTNPrefix]
Description (Modification: New parameters - PstnPrefix_SrcIPGroupID, PstnPrefix_DestHostPrefix, and PstnPrefix_SrcHostPrefix.) This ini file table parameter configures the routing of IP-to-Tel calls to Hunt Groups. The format of this parameter is as follows: [PSTNPrefix] FORMAT PstnPrefix_Index = PstnPrefix_DestPrefix, PstnPrefix_TrunkGroupId, PstnPrefix_SourcePrefix, PstnPrefix_SourceAddress, PstnPrefix_ProfileId, PstnPrefix_SrcIPGroupID, PstnPrefix_DestHostPrefix, PstnPrefix_SrcHostPrefix; [\PSTNPrefix] Where,

DestPrefix = Destination number prefix. TrunkGroupId = Hunt Group ID (1-99). SourcePrefix = Source number prefix. SourceAddress = Source IP address (obtained from the Contact header in the INVITE message). ProfileId = Optional IP Profile ID (1-4) that can be applied to each routing rule. SrcIPGroupID = Source IP Group (1-9) associated with the incoming IPto-Tel call. DestHostPrefix = Request URI host name prefix of the incoming INVITE message. SrcHostPrefix = From URI host name prefix of the incoming INVITE message.
For example: [PSTNPrefix] FORMAT PstnPrefix_Index = PstnPrefix_DestPrefix, PstnPrefix_TrunkGroupId, PstnPrefix_SourcePrefix, PstnPrefix_SourceAddress, PstnPrefix_ProfileId, PstnPrefix_SrcIPGroupID, PstnPrefix_DestHostPrefix, PstnPrefix_SrcHostPrefix; PstnPrefix 0 = 100, 1, 200, *, 0, 2, , ; PstnPrefix 1 = *, 2, *, , 1, 3, acl, joe; [\PSTNPrefix] Notes:

This parameter can include up to indices. To support the In-Call Alternative Routing feature, you can use two entries that support the same call, but assigned with a different HuntGroup. The second entry functions as an alternative selection if the first rule fails as a result of one of the release reasons listed in the AltRouteCauseIP2Tel table. Selection of Hunt Groups (for IP-to-Tel calls) is according to destination number, source number,and source IP address. The source IP address (SourceAddress) can include the 'x' wildcard to represent single digits. For example: 10.8.8.xx represents all IP addresses between 10.8.8.10 and 10.8.8.99. The source IP address (SourceAddress) can include the asterisk ('*') wildcard to represent any number between 0 and 255. For example, 10.8.8.* represents all addresses between 10.8.8.0 and 10.8.8.255. 32 Document #: LTRT-65610

SIP Release Notes
SIP Release Notes
Web / [ini] File Parameter Name SIP Transport Type [SIPTransportType]
Description (Modification: Addition of SAS note.) Determines the default transport layer for outgoing SIP calls initiated by the device.

[0] UDP (default) [1] TCP [2] TLS (SIPS) It's recommended to use TLS for communication with a SIP Proxy and not for direct device-to-device communication. The value of this parameter is also used by the SAS application as the default transport layer for outgoing SIP calls.
Notes:

SAS Registration Time (Modification: Corresponding Web interface parameter.) [SASRegistrationTime] Determines the value of the SIP Expires header that is sent in a 200 OK response to an incoming REGISTER message when in SAS 'Emergency Mode'. The valid range is 0 to 2,000,000. The default value is 20. Registrar IP Address [RegistrarIP] (Modification: Addition of a note.) IP address (or FQDN) and optionally, port number of the SIP Registrar server. The IP address is in dotted-decimal notation, e.g., 201.10.8.1:<5080>. Notes:

If not specified, the REGISTER request is sent to the primary Proxy server (refer to 'Proxy IP address' parameter). When a port number is specified, DNS NAPTR/SRV queries aren't performed, even if DNSQueryType is set to 1 or 2. If the RegistrarIP is set to an FQDN and is resolved to multiple addresses, the device also provides real-time switching (hotswap mode) between different Registrar IP addresses (IsProxyHotSwap is set to 1). If the first Registrar doesn't respond to the REGISTER message, the same REGISTER message is sent immediately to the next Proxy. EnableProxyKeepAlive must be set to 0 for this logic to apply. When a specific Transport Type is defined using RegistrarTransportType, a DNS NAPTR query is not performed even if DNSQueryType is set to 2.
Version 5.4
33
May 2008
MediaPack Series Web / [ini] File Parameter Name DNS Query Type [DNSQueryType]
Description (Modification: Limitation on the usage of specific transport types.) Enables the use of DNS Naming Authority Pointer (NAPTR) and Service Record (SRV) queries to resolve Proxy and Registrar servers and to resolve all domain names that appear in the Contact and Record-Route headers.

[0] A-Record = A-Record (default) [1] SRV = SRV [2] NAPTR = NAPTR
If set to A-Record [0], no NAPTR or SRV queries are performed. If set to SRV [1] and the Proxy / Registrar IP address parameter, Contact / Record-Route headers, or IP address defined in the Routing tables contains a domain name, an SRV query is performed. The device uses the first host name received from the SRV query. The device then performs a DNS Arecord query for the host name to locate an IP address. If set to NAPTR [2], an NAPTR query is performed. If it is successful, an SRV query is sent according to the information received in the NAPTR response. If the NAPTR query fails, an SRV query is performed according to the configured transport type. If the Proxy / Registrar IP address parameter, the domain name in the Contact / Record-Route headers, or the IP address defined in the Routing tables contains a domain name with port definition, the device performs a regular DNS A-record query. If a specific Transport Type is defined, a NAPTR query is not performed. Note: To enable NAPTR/SRV queries for Proxy servers only, use the parameter ProxyDNSQueryType. Proxy DNS Query Type (Modification: Limitation on the usage of specific transport types.) [ProxyDNSQueryType] Enables the use of DNS Naming Authority Pointer (NAPTR) and Service Record (SRV) queries to discover Proxy servers.

[0] A-Record = A-Record (default) [1] SRV = SRV [2] NAPTR = NAPTR
If set to A-Record [0], no NAPTR or SRV queries are performed. If set to SRV [1] and the Proxy IP address parameter contains a domain name without port definition (e.g., ProxyIP = domain.com), an SRV query is performed. The SRV query returns up to four Proxy host names and their weights. The device then performs DNS A-record queries for each Proxy host name (according to the received weights) to locate up to four Proxy IP addresses. Therefore, if the first SRV query returns two domain names, and the A-record queries return two IP addresses each, no additional searches are performed. If set to NAPTR [2], an NAPTR query is performed. If it is successful, an SRV query is sent according to the information received in the NAPTR response. If the NAPTR query fails, an SRV query is performed according to the configured transport type. If the Proxy IP address parameter contains a domain name with port definition (e.g., ProxyIP = domain.com:5080), the device performs a regular DNS A-record query.
SIP Release Notes
34
SIP Release Notes
Description If a specific Transport Type is defined, a NAPTR query is not performed. Note: When enabled, NAPTR/SRV queries are used to discover Proxy servers even if the parameter DNSQueryType is disabled.
Hook-Flash Option [HookFlashOption]
(Modification: New option [5] INFO.) The supported hook-flash Transport Type (method by which hook-flash is sent and received).

[0] Not Supported = Hook-Flash indication isn't sent (default). [1] INFO = Send proprietary INFO message with Hook-Flash indication. [4] RFC 2833 [5] INFO (Lucent) = Send proprietary INFO message with Hook-Flash indication. The FXO interfaces support the receipt of RFC 2833 Hook-Flash signals. The FXS interfaces send Hook-Flash signals only if EnableHold is set to 0.
Notes:

[SIPSRequireClientCer (Modification: Parameter description and additional note.) tificate] Determines the device's behavior when acting as a server for TLS connections.

[0] = The device does not request the client certificate (default). [1] = The device requires receipt and verification of the client certificate to establish the TLS connection. The SIPS certificate files can be changed using the parameters HTTPSCertFileName and HTTPSRootFileName. This parameter cannot be changed on-the-fly and requires a device reset.
Notes:

Max. Flash-Hook (Modification: Addition of usage option for FXS interfaces.) Detection Period [msec] Defines the hook-flash period (in msec) for both analog and IP sides. For [FlashHookPeriod] the IP side, it defines the hook-flash period that is reported to the IP. For the analog side, it defines the following:

FXS interfaces: Maximum hook-flash detection period. A longer signal is considered an off-hook or on-hook event. FXS interfaces: Hook-flash generation period upon detection of a SIP INFO message containing a hook-flash signal. FXO interfaces: Hook-flash generation period.
The valid range is 25 to 1,500. The default value is 700. Note: For FXO interfaces, a constant of 100 msec must be added to the required hook-flash period. For example, to generate a 450 msec hookflash, set this parameter to 550.
Version 5.4
35
May 2008
Description
[TrunkGroupSettings] (Modification: New parameter TrunkGroupSettings_ServingIPGroup.) This ini file table parameter defines rules for port allocation per Hunt Group. If no rule exists, the global rule defined by the parameter ChannelSelectMode takes effect. The format of this parameter is as follows: [TrunkGroupSettings] FORMAT TrunkGroupSettings_Index = TrunkGroupSettings_TrunkGroupId, TrunkGroupSettings_ChannelSelectMode, TrunkGroupSettings_RegistrationMode, TrunkGroupSettings_GatewayName,TrunkGroupSettings_ContactUser, TrunkGroupSettings_ServingIPGroup; [\TrunkGroupSettings] Where,

TrunkGroupId = Hunt Group ID number. ChannelSelectMode = Channel select mode for the HuntGroup. Available values are identical to those defined by the ChannelSelectMode parameter. RegistrationMode = Registration mode for the HuntGroup (Per Endpoint [0], Per Gateway [1], or Do Not Register [4]). If not configured [-1], the value of AuthenticationMode is used. GatewayName = sipgatewayname used as a hostname in the From header in INVITE and REGISTER messages. If not configured, the sipgatewayname parameter is used. ContactUser = User part in contact URI in INVITE, and in From, To and Contact headers in REGISTER. ServingIPGroup = Serving IP Group ID to where INVITE messages initiated by the Hunt Group endpoints are sent.

For example: [TrunkGroupSettings] TrunkGroupSettings 0 = 1, 0, 5, audiocodes, user, 1; TrunkGroupSettings 1 = 2, 1, 0, localname, user1, 2; [\TrunkGroupSettings] Note: This parameter can include up to 24 indices.
SIP Release Notes
36
SIP Release Notes
Web / [ini] File Parameter Name [ProxyIP]
Description (Modification: New parameters ProxyIp_TransportType and ProxyIp_ProxySetId.) This ini file table parameter configures the Proxy Set ID table for configuring up to six Proxy Sets, each with up to five Proxy server IP addresses. The format of this parameter is as follows: [ProxyIP] FORMAT ProxyIp_Index = ProxyIp_IpAddress, ProxyIp_TransportType, ProxyIp_ProxySetId; [\ProxyIP] Where,

IpAddress = Proxy server's IP address. TransportType = Not configured (-1) - the global parameter SIPTransportType type is used; UDP (0); TCP (1); TLS (2). ProxySetId = ID of the Proxy Set.
For example: [ProxyIP] FORMAT ProxyIp_Index = ProxyIp_IpAddress, ProxyIp_TransportType, ProxyIp_ProxySetId; ProxyIp 0 = 10.33.37.77, -1, 0; ProxyIp 1 = 10.8.8.10, 0, 2; ProxyIp 2 = 10.8.8.40, -1, 1; ProxyIp 3 = 10.5.6.7, -1, 1; [\ProxyIP] Notes:

This parameter can include up to 30 indices (0-29). For assigning various attributes (such as Proxy Load Balancing) to each Proxy Set ID, refer to the ini file parameter ProxySet.
Version 5.4
37
May 2008
MediaPack Series
1.7
Obsolete Parameters
The table below lists parameters from the previous release that are now obsolete. Table 1-3: Release 5.4 Obsolete Web / [ini ] File Parameters Web / [ini] File Parameter Name Description This parameter is obsolete; instead, use ProxySet. This parameter is obsolete for this release. This parameter is obsolete for this release. This parameter is obsolete for this release. This parameter is obsolete for this release. This parameter is obsolete for this release. This parameter is obsolete for this release. This parameter is obsolete for this release.
[ProxyIP] [EnablePPPoE] [PPPoEUserName] [PPPoEPassword] [PPPoEServerName] [PPPoEStaticIPAddress] [PPPoERecoverIPAddress] [PPPoERecoverSubnetMask]
[PPPoERecoverDfGWAddress] This parameter is obsolete for this release. [PPPoELCPEchoEnable] This parameter is obsolete for this release.
SIP Release Notes
38
SIP Release Notes
2. Supported Features
2
2.1
2.1.1
Supported Features
SIP Features
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: 10453@212.179.22.229 Seq: 1 REGISTER Expires: 3600 Contact: sip:GWRegistrationName@212.179.22.229 Content-Length: 0 The "servername" string is defined according to the following rules: 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).

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.
Version 5.4
39
May 2008
MediaPack Series 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 Basic or Digest methods. 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. Supports Tel URI (Uniform Resource Identifier) according to RFC 2806 bis.

SIP Release Notes
40
SIP Release Notes

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 1225556@10.1.2.4, where 122556 is the phone number of the source or destination) or sip:phone_number@domain name, such as 122556@myproxy.com. 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 recepit of a REFER method outside of a dialog. Support RFC 4458, SIP URIs for Applications such as Voicemail 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). Support RFC 3455, Private Header (P-Header) Extensions to SIP for the 3rdGeneration Partnership Project (3GPP) [Partial].

Version 5.4
41
May 2008
MediaPack Series Support RFC 4235, An INVITE-Initiated Dialog Event Package for SIP [Partial]. Support RFC 3680, A SIP Event Package for Registrations.
2.1.2
Unsupported SIP Features

The following SIP features are not supported: MESSAGE method Preconditions (RFC 3312) SDP - Simple Capability Declaration (RFC 3407) S/MIME
2.2
SIP Compliance Tables

The SIP device complies with RFC 3261, as shown in the following subsections.
2.2.1
SIP Functions
The device supports the following SIP Functions: Table 2-1: Supported SIP Functions Function Supported Yes Yes Third-party only tested with, amongst others, Ubiquity, Delta3, Microsoft, 3Com, BroadSoft, Snom and Cisco Proxies) Third-party Third-party Yes Third-party
User Agent Client (UAC) User Agent Server (UAS) Proxy Server Redirect Server Registrar Server Event Publication Agent (EPA) Event State Compositor (ESC)
2.2.2
SIP Methods
The device supports the following SIP Methods: Table 2-2: Supported SIP Methods
Method INVITE ACK BYE
Supported Yes Yes Yes
Comments
SIP Release Notes
42
SIP Release Notes
Method CANCEL REGISTER REFER NOTIFY INFO OPTIONS PRACK UPDATE PUBLISH SUBSCRIBE
Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Send only Send only
Comments
Inside and outside of a dialog
2.2.3
SIP Headers
The device supports the following SIP Headers: Table 2-3: Supported SIP Headers Header Field Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes No Yes
Accept AcceptEncoding Alert-Info Allow Also Asserted-Identity Authorization Call-ID Call-Info Contact Content-Disposition Content-Encoding Content-Length Content-Type Cseq Date Diversion Encryption Expires
Version 5.4
43
May 2008
MediaPack Series Header Field Fax From History-Info Join Max-Forwards Messages-Waiting MIN-SE Organization P-Associated-URI P-Asserted-Identity P-Charging-Vector P-Preferred-Identity Priority Proxy- Authenticate Proxy- Authorization Proxy- Require Prack Reason Record- Route Refer-To Referred-By Replaces Require Remote-Party-ID Response- Key Retry-After Route Rseq Session-Expires Server Service-Route SIP-If-Match Subject Supported Target-Dialog Timestamp SIP Release Notes 44 Supported Yes Yes Yes Yes Yes Yes Yes No Yes (Receive Only) Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Document #: LTRT-65610
SIP Release Notes
Header Field To Unsupported User- Agent Via Voicemail Warning WWW- Authenticate
Supported Yes Yes Yes Yes Yes Yes Yes
2.2.4
SDP Headers
The device supports the following SDP Headers: Table 2-4: Supported SDP Headers SDP Header Element Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes
v - Protocol version o - Owner/ creator and session identifier a - Attribute information c - Connection information d - Digit m - Media name and transport address s - Session information t - Time alive header b - Bandwidth header u - Uri Description Header e - Email Address header i - Session Info Header p - Phone number header y - Year
2.2.5
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 45 May 2008
Version 5.4
MediaPack Series
2.2.5.1
1xx Response Information Responses

Table 2-5: Supported 1xx SIP Responses
1xx Response 100 Trying
Supported Yes
Comments The SIP device generates this response upon receiving a Proceeding message from ISDN or immediately after placing a call for CAS signaling. The SIP device generates this response for an incoming INVITE message. Upon receiving this response, the device waits for a 200 OK response. The SIP 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. The SIP 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. The SIP device generates this response if the Early Media feature is enabled and if the device plays a Ringback tone to IP
180
Ringing
Yes
181
Call is Being Forwarded
Yes
182
Queued
Yes
183
Session Progress
Yes
2.2.5.2
2xx Response Successful Responses

2xx Response 200 202 OK Accepted
Supported Yes Yes
Comments ---
2.2.5.3
3xx Response Redirection Responses

3xx Response 300 301 302 Multiple Choice Moved Permanently Moved Temporarily
Supported Yes Yes Yes
Comments The device responds with an ACK, and then resends the request to the first new address in the contact list. The device responds with an ACK, and then resends the request to the new address. The SIP 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. The device responds with an ACK, and then resends the request to a new address. The device responds with an ACK, and then resends the request to a new address. 46 Document #: LTRT-65610
305 380
Use Proxy Alternate Service
Yes Yes
SIP Release Notes
SIP Release Notes
2.2.5.4
4xx Response Client Failure Responses

4xx Response 400 Bad Request
Supported Yes
Comments 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. Authentication support for Basic and Digest. Upon receiving this message, the device issues a new request according to the scheme received on this response. 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. 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. The SIP 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. 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. 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. Authentication support for Basic and Digest. Upon receiving this message, the device issues a new request according to the scheme received on this response. 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. 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. 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. 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. 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. 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. 47 May 2008
401
Unauthorized
Yes
402
Payment Required Forbidden
Yes
403
Yes
404
Not Found
Yes
405
Method Not Allowed Not Acceptable
Yes
406
Yes
407
Proxy Authentication Required
Yes
408 Request Timeout
Yes
409
Conflict
Yes
410
Gone
Yes
411
Length Required
Yes
413
Request Entity Too Large Unsupported Media
Yes
415
Yes
Version 5.4
MediaPack Series 4xx Response 420 Bad Extension Supported Yes Comments 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. The device does not generate this response. On reception of this message the device uses the value received in the MinExpires header as the registration time. 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. 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. 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. 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. 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. 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. 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. 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. This response indicates that the initial request is terminated with a BYE or CANCEL request. 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. When acting as a UAS: the device sent a re-INVITE on an established session and is still in progress. If it receives a reINVITE 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.
423 Interval Too Brief
Yes
433
Anonymity Disallowed
Yes
480
Temporarily Unavailable Call Leg/Transaction Does Not Exist Loop Detected
Yes
481
Yes
482
Yes
483
Too Many Hops
Yes
484
Address Incomplete Ambiguous
Yes
485
Yes
486
Busy Here
Yes
487 488
Request Canceled Not Acceptable
Yes Yes
491 Request Pending
Yes
SIP Release Notes
48
SIP Release Notes
2.2.5.5
5xx Response Server Failure Responses

Table 2-9: Supported 5xx SIP Responses 5xx Response Comments
500 501 502 503 504 505
Internal Server Error Not Implemented Bad gateway Service Unavailable Gateway Timeout Version Not Supported 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.
2.2.5.6
6xx Response Global Responses

Table 2-10: Supported 6xx SIP Responses 6xx Response Comments
600 603 604 606
Busy Everywhere Decline Does Not Exist Anywhere Not Acceptable Upon receipt of any of these Responses, the device releases the call, sending an appropriate release cause to the PSTN side.
Version 5.4
49
May 2008
MediaPack Series
Reader's Notes
SIP Release Notes
50
SIP Release Notes
3. Known Constraints
Known Constraints
This section lists known constraints in Release 5.4. Note: Due to the improved ini file representation for tables, it's not possible to load an ini file that was used by a device running software version 5.2 or 5.4 to a device using an earlier version (e.g. 5.0). This can result in an invalid configuration.
3.1
SIP Constraints
This release includes the following known SIP constraints: 1. Channel parameters such as voice/DTMF gain and jitter buffer are collectively configured in the ini file per device (not per call). By using Profiles, this limitation can be overcome. The number of RTP payloads packed in a single G.729 packet (M channel parameter) is limited to 5.
2.
3.2
Gateway Constraints
This release includes the following known gateway constraints: 1. The device attempts to access the wrong TFTP server when the IniFileUrl parameter specifies a TFTP URL. It is possible to work around this problem by resetting the device (using the Web interface) after the TFTP error occurs. Debug Recording: 3. 4. Only one IP target is allowed. Maximum of 50 trace rules are allowed simultaneously. Maximum of 5 media stream recordings are allowed simultaneously.
2.
VLAN Pass-Through mode is not supported. In some cases, when the spanning-tree algorithm is enabled on the external Ethernet switch port connected to the device, the external switch blocks traffic entering and exiting the device for some time after the device is reset. This may cause a loss of important packets (such as BootP and TFTP requests) which in turn may cause the device to fail to start up. A possible workaround for this issue is to set the parameter BootPRetries to 5, causing the device to issue 20 BootP requests for 60 seconds. A second workaround is to disable the spanning-tree algorithm on the port of the external switch that is connected to the device. 10Base-T Half-Duplex is not supported (only 10/100Base-T Full Duplex and 100BaseT Half-Duplex are supported). Configuring the device to auto-negotiate mode while the opposite port is set manually to full-duplex (either 10Base-T or 100Base-TX) is invalid. It is also invalid to set the device to one of the manual modes while the opposite port is configured differently. It is recommended to use full-duplex connections instead of half-duplex, and 100BaseTX instead of 10Base-T (due to the larger bandwidth). ThroughPacket RTP Multiplexing is not supported with VLAN configurations.
5. 6.
7.
Version 5.4
51
May 2008
MediaPack Series 8. 9. PPPoE is not supported. NTT caller ID type 2 constraints: The NTT standard describes the CallerID type 2 generation as a sequence of an incoming-call signal, 'C' and 'D' DTMFs, and FSK modulated Data. Generation of the incoming call signal remains the responsibility of the application, but 'C', 'D', and the FSK are generated by the supplied service. The signal can be generated using the UDT signal generation mechanism. Prior to the detection of NTT CallerID type 2, there are two DTMF ('C' and 'D') detections which remain unscreened.
10. The 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 inter-digit information lost. 11. Transcoding is not supported with coder frame sizes other than the default size (refer to SampleBasedCodersRTPPacketInterval). 12. The resolution of the duration of digits On and Off time when dialing to the IP side using RFC 2833 relay is dependent on the basic frame size of the coder being used. 13. When using a sample interval of 10 or 5 msec, the channel capacity may be reduced. 14. When using m-factor values greater than 8, you must set jitter buffer optimization to 13 to cancel any jitter optimization and avoid under running condition. 15. When using SRTP, channel capacity is reduced. Contact AudioCodes for more details. 16. If a voice prompt is used before a silence in a Voice Prompt Sequence, the end of the voice prompt is cut off by up to 80 msec. To avoid this, a silence period of 80 msec should be added to the end of the specific voice prompt. 17. When using SRTP, the number of basic codec frames per RTP packet cannot be greater than 1. In addition, the RTP Redundancy (RFC 2198) feature cannot be activated. 18. The DJBufOptFactor parameter cannot be set to 13 if the channel is configured to operate with Silence Compression enabled. 19. Transparent With Events Bell modem Transport Type is not supported. 20. Flash-burning control for specific files (BurnCASFile, BurnCallProgressToneFile, BurnVXMLFile, BurnVoicePromptsFile) is no longer supported. The new SaveConfiguration parameter must be used instead. 21. Date and time should be set after each device reset, unless Network Time Protocol (NTP) is used. 22. Daylight Savings Time is not supported. 23. The following constraints apply when defining coders via the ini file. Coder names are case-sensitive. Don't use obsolete coder names (e.g., g729_AnnexB, g7231r53) with the improved coder interface. When an invalid packetization time is used, the coder definition is disregarded. When an invalid rate is used for dynamic-rate coders, the coder definition is disregarded.
24. The device supports only symmetrical coders - the same coder is used for transmit and receive (though different ptime is supported). SIP Release Notes 52 Document #: LTRT-65610
SIP Release Notes
3. Known Constraints
25. The 'Transparent' coder doesn't use DSP resources, therefore, the DSP functionality is off (i.e., DTMF detection, silence detection, etc.) and a reset is needed before switching to a different coder. 26. It is highly recommended to use 100Base-T switches. Use of 10Base-T LAN hubs should be avoided. 27. Static NAT is not supported for local IP calls. 28. Tables that use the improved ini file representation cant be burned to flash memory as Client Defaults.
3.3
Web Constraints
This release includes the following known Web constraints: 1. On the 'Software Upgrade Wizard' page of the Web interface, the Software Upgrade process should be completed prior to clicking the Back button. Clicking the Back button before the wizard completes causes a display distortion. When downloading a TAR file from the 'Auxiliary Files' page of the Web interface, the progress bar may not display the correct status; however, the number of bytes is displayed correctly. The following Web interface's configuration pages do not support the Scenario mode: 4. Web User Accounts Web & Telnet Access List Regional Settings
2.
3.
For users who have 'Read Only' access to the Web interface, the 'Read Only Mode' string text does not appear in bold format on the following pages: Tel to IP Routing Table, Trunk Group Settings, SNMP Community String and SNMP Trap Destinations. The Routing table can be configured in the Web interface, but the ini file is not updated with the new settings. This version does not support screen resolution 1152 x 864. On the 'IP Settings' page, when selecting a 'multiple' or 'dual' value from the 'IP Networking Mode' field, the 'DHCP' field is incorrectly enabled. Not all parameters can be changed on-the-fly in the Web interface. Parameters that symbol. To change these can't be changed on-the-fly are noted with the lightning parameters, reset the device using the Web interface's Reset button. When changing device parameters in the Web interface, the new parameters are permanently stored in flash memory only after the device is reset from the Web or after the BURN button is clicked in the 'Maintenance Actions' page.
5. 6. 7. 8.
9.
10. The number of fax calls displayed in the fields 'Attempted Fax Calls Counter' and 'Successful Fax Calls Counter' in the 'Calls Count' pages may not be accurate. 11. In the 'Coders' and 'Coder Group Settings' pages, the voice quality is reduced when G.729 is used with ptime 120, and G.723 is used with ptime 150. Therefore, using these ptimes is not recommended. 12. When loading an ini file in the Web interface, the 'swwd' messages appears.
Version 5.4
53
May 2008
MediaPack Series
3.4
SNMP Constraints
This release includes the following known SNMP constraints: 1. 2. 3. 4. In the ipCidrRouteTable, new rows cannot be added and rows that were previously deleted using the Web interface, cannot be deleted. When configuring the acSysInterfaceTable using SNMP or the Web interface, validation is only performed after device reset. When attempting to enable Telnet using SNMP, a fail notification is displayed despite the operation being successful. When defining or deleting SNMPv3 users, the v3 trap user must not be the first or last to be defined. If there are no non-default v2c users, this results in a loss of SNMP contact with the device. The SNMPv3 users table returns the line removed notice when adding a new row to an active row index. After adding an empty line to the SNMPV3 table, it's impossible to delete it or add new lines. The default values created in an IPSec configuration table are incorrect. The user should override the default values before activating the new row. The acBoardConfigurationError alarm trap, generated as a result of a configuration error, does not clear. The following RTP MIB objects are not supported: rtpRcvrSRCSSRC, rtpRcvrSSRC, rtpSenderSSRC, rtpRcvrLostPackets, rtpRcvrPackets, rtpSenderPackets, rtpRcvrOctets, and rtpSenderOctets.
5. 6. 7. 8. 9.
10. An Ethernet link trap is sent before the link is up - manager does not receive clear. This occurs because a spanning tree algorithm is being calculated in the Ethernet switch. 11. The following encryptions types are currently supported (for SNMP v3 users only): Authentication protocol: MD5 and SHA Privacy protocol: DES and AES128
12. The range of the faxModemRelayVolume MIB object is incorrect. Instead of 0 to 15, it should be -18 to -3, corresponding to an actual volume of -18.5 dBm to -3.5 dBm. 13. Only one SNMP manager can access the device simultaneously.
3.5
CLI Constraints
This release includes the following known CLI constraint: 1. When connecting to a device using Telnet (CLI), Syslog messages do not appear by default. The show log command must be used to enable this feature.
SIP Release Notes
54
SIP Release Notes
4. Resolved Constraints
Resolved Constraints
The following constraints from previous releases have been resolved in Release 5.4: 1. 2. Web Interface: When configuring the RadiusAuthServerIp parameter with a nonexistent server IP, the BehaviorUponRadiusTimeout parameter value is ignored. Web Interface: Users cannot enter a username/password of 8 characters.
Version 5.4
55
May 2008
MediaPack Series
Reader's Notes
SIP Release Notes
56
SIP Release Notes
5. Earlier Releases
Earlier Releases
Details of previous releases can be found in the Release Notes of Version 5.2, published by AudioCodes on May-24-2007.
Version 5.4
57
May 2008
Release Notes
Version 5.4
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LTRT-65610 MP-11x and MP-124 SIP Release Notes Ver 5.4

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LTRT-65610 MP-11x and MP-124 SIP Release Notes Ver 5.4

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Release Notes

SIP Release Notes

1.2 1.3 1.4 1.5 1.6 1.7

Supported Features ..........................................................................................39

SIP Compliance Tables.......................................................................................... 42

Known Constraints ...........................................................................................51

Resolved Constraints .......................................................................................55 Earlier Releases ................................................................................................57

SIP Release Notes

SIP Release Notes

Abbreviations and Terminology

SIP Release Notes

SIP Release Notes

1. What's New in Release 5.4

What's New in Release 5.4

Supported Hardware Platforms

Support of the Existing Hardware Platforms

MP-124/FXS providing 24 analog FXS interfaces

Hardware Platforms No Longer Supported

General Gateway New Features

Increased URL Length for CmpFileURL Parameter: MP-11x FXS FXO

SIP Release Notes

SIP Release Notes

1. What's New in Release 5.4

Automatic Configuration URL Setup in Voice Menu: MP-11x FXS FXO

MediaPack Series 9. Additional CDR Fields: MP-11x FXS FXO

SIP Release Notes

SIP Release Notes

1. What's New in Release 5.4

13. Reject Anonymous Calls: MP-124 MP-11x FXS FXO

Parameter Value Not Relevant 0 4 3 or 5

Registered Number 12341234 12341234 12341234 12341234

Dialed Number 12341234 1234 1234 1234

Match? Yes No Yes No

SIP Release Notes

SIP Release Notes

1. What's New in Release 5.4

SIP New Features

Relevant parameter: EnablePAssociatedURIHeader.

SIP Release Notes

SIP Release Notes

1. What's New in Release 5.4

Set OPTIONS Request-URI User-Part: MP-11x FXS FXO

MediaPack Series 7. TCP and TLS Enhancements: MP-11x FXS FXO

SIP Release Notes

SIP Release Notes

1. What's New in Release 5.4

Web and SNMP New Features

Web / [ini] File Parameter Name IPSec Mode [IPSecMode]

Description Defines the IPSec mode of operation.

[0] Transport (Default) [1] Tunneling

[0] Disable (default). [1] Enable.

SIP Release Notes

SIP Release Notes

1. What's New in Release 5.4

Web / [ini] File Parameter Name

Enable = accept [0] (default) or reject [1] incoming anonymous calls.

MediaPack Series Web / [ini] File Parameter Name

[1] TCP [2] TLS

SIP Release Notes

SIP Release Notes

1. What's New in Release 5.4

Web / [ini] File Parameter Name

[0] UDP [1] TCP [2] TLS

[0] Disable (default). [1] Enable.