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

Document #: LTRT-65610 May 2008

Release Notes

Version 5.4

Version 5.4 3 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 New Products Introduced in this Release..................................................................7 1.1.2 Support of the Existing Hardware Platforms..............................................................7 1.1.3 Hardware Platforms No Longer Supported ...............................................................7

1.2 General Gateway New Features..............................................................................8 1.3 SIP New Features ..................................................................................................14 1.4 Web and SNMP New Features ..............................................................................17 1.5 New Parameters ....................................................................................................18 1.6 Modified Parameters ..............................................................................................28 1.7 Obsolete Parameters .............................................................................................38

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

2.1 SIP Features ..........................................................................................................39 2.1.1 Supported SIP Features ..........................................................................................39 2.1.2 Unsupported SIP Features ......................................................................................42

2.2 SIP Compliance Tables..........................................................................................42 2.2.1 SIP Functions ..........................................................................................................42 2.2.2 SIP Methods ............................................................................................................42 2.2.3 SIP Headers ............................................................................................................43 2.2.4 SDP Headers...........................................................................................................45 2.2.5 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

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

3.1 SIP Constraints ......................................................................................................51 3.2 Gateway Constraints..............................................................................................51 3.3 Web Constraints.....................................................................................................53 3.4 SNMP Constraints..................................................................................................54 3.5 CLI Constraints ......................................................................................................54

4 Resolved Constraints .......................................................................................55

5 Earlier Releases ................................................................................................57

SIP Release Notes 4 Document #: LTRT-65610

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 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, CTI², 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 [email protected].

Abbreviations and Terminology

Each abbreviation, unless widely used, is spelled out in full when first used. Only industry-standard terms are used throughout this manual. Hexadecimal notation is indicated by 0x preceding the number.

http://www.audiocodes.com/

mailto:[email protected]


MediaPack Series

Related Documentation

Document # Manual Name

LTRT-523xx (where xx denotes the document version)

Product Reference Manual

LTRT-654xx MP-11x and MP-124 SIP User's Manual

LTRT-598xx MP-11x and MP-124 SIP-MGCP Installation Manual

LTRT-665xx CPE Configuration Guide for IP Voice Mail

Notes:

• Throughout this manual, the terms MediaPack or device refer to the MP-124, 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 1. What's New in Release 5.4

1 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 Supported Hardware Platforms

1.1.1 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-124/FXS providing 24 analog FXS interfaces

1.1.3 Hardware Platforms No Longer Supported Not applicable.


MediaPack Series

1.2 General Gateway New Features The device supports the following new gateway features:

1. IPSec Tunneling Mode:


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):


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:


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:


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.



5. Automatic Configuration URL Setup in Voice Menu:


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:


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:

• 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 µ-Law compatible.

Relevant parameters: CoderName; CoderName_ID.

7. Special DTMF Representation:


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:


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.


MediaPack Series

9. Additional CDR Fields:


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:


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:


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:


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.



13. Reject Anonymous Calls:


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:


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:


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.


MediaPack Series

The table below provides a few examples of the SAS short numbering feature:

Parameter Value Registered Number Dialed Number Match?

Not Relevant 12341234 12341234 Yes

0 12341234 1234 No

4 12341234 1234 Yes

3 or 5 12341234 1234 No

Note: This feature operates in 'Emergency' mode only.

Relevant parameter: SASShortNumberLength.

16. SAS TCP/TLS:


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:


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:


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.



19. Configurable Transport Type for Servers and IP Addresses:


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):


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:


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:


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.


MediaPack Series

1.3 SIP New Features The device supports the following new SIP features:

1. Retry-After SIP Header:


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:


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]:


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.



4. RFC 4235 (An INVITE-Initiated Dialog Event Package for SIP) [Partial]:


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:

• 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>).

5. RFC 3680 (A SIP Event Package for Registrations):


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:

• 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.

6. Set OPTIONS Request-URI User-Part:


It is now possible to define the Request-URI User-Part value, which is used in outgoing SIP OPTIONS requests.

Relevant parameter: OPTIONSUserPart.


MediaPack Series

7. TCP and TLS Enhancements:


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:


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.



1.4 Web and SNMP New Features The device supports the following new Web and SNMP features:

1. Web Interface Scenario Feature:


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:


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:


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:


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.


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 Description

IPSec Mode [IPSecMode]

Defines the IPSec mode of operation. [0] Transport (Default) [1] Tunneling

Remote Tunnel IP Address [IPSecPolicyRemoteTunnelIPAddress]

Defines the IP address of the remote IPSec tunneling device.

Note: This parameter is only available if the parameter IPSecMode is set to Tunneling (1).

Remote Subnet Mask [IPsecPolicyRemoteSubnetMask]

Defines the subnet mask of the remote IPSec tunneling device. The default value is 255.255.255.255 (i.e., host-to-host IPSec tunnel).

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 Representation [UseDigitForSpecialDTMF]

Defines the representation for ‘special’ digits (‘*’ and ‘#’) that are used for out-of-band DTMF signaling (using SIP INFO/NOTIFY). [0] Special = Uses the strings ‘*’ and ‘#’ (default). [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).

[KeyBlindTransferAddPrefix]

Determines whether the device adds the Blind Transfer code (KeyBlindTransfer) to the dialed destination number. [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.





[RejectAnonymousCallPerPort_ID]

Determines whether the device rejects incoming anonymous calls on FXS interfaces. 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 [KeyRejectAnonymousCall]

Keypad sequence that activates the reject anonymous call option, whereby the device rejects incoming anonymous calls. After the sequence is pressed, a confirmation tone is heard.


Deactivate [KeyRejectAnonymousCallDeact]

Keypad sequence that de-activates the reject anonymous call option. After the sequence is pressed, a confirmation tone is heard.


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 Re-INVITE 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’


Emergency Calls Reanswer Timeout [EmergencyRegretTimeout]

Determines the time (in minutes) that the device waits before tearing-down 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.



MediaPack Series


Short Number Length [SASShortNumberLength]

Determines the length of the suffix used for Stand-Alone Survivability (SAS) Short Numbering support. The dialed number is compared to the registered number. The registered (full) and dialed (full or short) numbers 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.


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.


Alt Routing Tel to IP Connectivity Method [AltRoutingTel2IPConnMethod]

Determines the method used by the device for periodically querying the connectivity status of a destination IP address. [0] ICMP Ping (default) = Internet Control Message Protocol (ICMP)

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 Type [MWIServerTransportType]

Determines the transport layer used for outgoing SIP dialogs initiated by the device to the MWI Server. [-1] Not Configured (default) [0] UDP [1] TCP [2] TLS

Note: When set to ‘Not Configured (-1)’, the value of the parameter SIPTransportType is used.




Registrar Transport Type [RegistrarTransportType]

Determines the transport layer used for outgoing SIP dialogs initiated by 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.

[OfferUnencryptedSRTCP]

Determines whether the device adds the UNENCRYPTED_SRTCP attribute to outgoing SDP messages when SRTP/SRTCP is enabled. [0] Disable (default). [1] Enable.

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).

[PrecedenceRingingType]

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 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 Restriction [EnableContactRestriction]

Determines whether or not the device sets the Contact header of outgoing INVITE requests to ‘anonymous’ for restricted calls. [0] = Disabled (default) [1] = Enabled

Enable P-Associated-URI Header [EnablePAssociatedURIHeader]

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

Note: P-Associated-URIs in registration responses is handled only if the device is registered per endpoint.


MediaPack Series


[OPTIONSUserPart] 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 (‘’).

[ReRegisterOnConnectionFailure]

Enables the device to perform SIP Re-Registration upon TCP/TLS connection failure. [0] Disable (default). [1] Enable.

[TLSReHandshakeInterval]

Defines the time interval (in minutes) between TLS Re-Handshake operations initiated by the device. The interval range is 0 to 1,500 minutes. The default is 0 (i.e., no TLS Re-Handshake).

[PeerHostNameVerificationMode]

Determines whether the device verifies the Subject Name of a remote certificate when establishing TLS connections. [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. [0] Disable (default). [1] Enable.

Note: If Subject Name verification is necessary, the parameter PeerHostNameVerificationMode must be used as well.




[TLSRemoteSubjectName]

Defines the Subject Name that is compared with the name defined in the remote side certificate when establishing TLS connections.

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 Refer-To/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).


MediaPack Series


[Account] 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-to-IP 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.




[ZeroSDPHandling] 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 Pattern [DigitPatternDigitToIgnore]

A digit pattern that if received as Src (S) or Redirect (R) numbers is ignored and not added to that number. The valid range is a 25-character string.

[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.

[SyslogOutputMethod] Determines the method used for Syslog messages. [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.

[SourceNumberPreference]

Determines the SIP header used to determine the Source Number in incoming INVITE messages. “” (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.


MediaPack Series


[EnableSilenceSuppInSDP]

Determines the device's behavior upon receipt of SIP Re-INVITE messages that include the silencesupp:off attribute. [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 Voice-Band-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. [BlindTransferDisconnectTimeout]

Defines the duration (in milliseconds) for which the device waits for a disconnection from the Tel side after the Blind Transfer Code (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.

[CallPriorityMode] Enables Priority Calls handling. [0] Disable (default). [1] MLPP = Priority Calls handling is enabled.




ProxySet 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.


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


[CmpFileURL] (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




value is used. 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.


MediaPack Series


Enable Busy Out [EnableBusyOut]

(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.

Notes: 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. [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.




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).


MediaPack Series


[PSTNPrefix] (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 IP-

to-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.




SIP Transport Type [SIPTransportType]

(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)

Notes: 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.

SAS Registration Time[SASRegistrationTime]

(Modification: Corresponding Web interface parameter.)

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.


MediaPack Series


DNS Query Type [DNSQueryType]

(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 containsa 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 A-record 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[ProxyDNSQueryType]

(Modification: Limitation on the usage of specific transport types.)

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.




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. Notes: 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.

[SIPSRequireClientCertificate]

(Modification: Parameter description and additional note.)

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. Notes: 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.

Max. Flash-Hook Detection Period [msec][FlashHookPeriod]

(Modification: Addition of usage option for FXS interfaces.)

Defines the hook-flash period (in msec) for both analog and IP sides. For 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 hook-flash, set this parameter to 550.


MediaPack Series


[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.




[ProxyIP] (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.


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


[ProxyIP] This parameter is obsolete; instead, use ProxySet.

[EnablePPPoE] This parameter is obsolete for this release.

[PPPoEUserName] This parameter is obsolete for this release.

[PPPoEPassword] This parameter is obsolete for this release.

[PPPoEServerName] This parameter is obsolete for this release.

[PPPoEStaticIPAddress] This parameter is obsolete for this release.

[PPPoERecoverIPAddress] This parameter is obsolete for this release.

[PPPoERecoverSubnetMask] This parameter is obsolete for this release.

[PPPoERecoverDfGWAddress] This parameter is obsolete for this release.

[PPPoELCPEchoEnable] This parameter is obsolete for this release.


SIP Release Notes 2. Supported Features

2 Supported Features

2.1 SIP Features

2.1.1 Supported SIP Features The device supports the following main SIP features:

Reliable User Datagram Protocol (UDP) transport, with retransmissions.

Transmission Control Protocol (TCP) Transport layer.

SIPS using TLS.

T.38 real time Fax (using SIP). Note: If the remote side includes the fax maximum rate parameter in the SDP body of the INVITE message, the device returns the same rate in the response SDP.

Operates with Proxy or without Proxy, using an internal routing table.

Fallback to internal routing table if Proxy is not responding.

Supports up to 15 Proxy servers. If the primary Proxy fails, the device automatically switches to a redundant Proxy.

Supports domain name resolving using DNS NAPTR and SRV records for Proxy, Registrar and domain names that appear in the Contact and Record-Route headers.

Supports Load Balancing over Proxy servers using Round Robin or Random Weights.

Proxy or Registrar Registration, such as: REGISTER sip:servername SIP/2.0 VIA: SIP/2.0/UDP 212.179.22.229;branch=z9hG4bRaC7AU234 From: <sip:GWRegistrationName@sipgatewayname>;tag=1c29347 To: <sip:GWRegistrationName@sipgatewayname> Call-ID: [email protected] Seq: 1 REGISTER Expires: 3600 Contact: sip:[email protected] Content-Length: 0

The "servername" string is defined according to the following rules:

• 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.


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.



Supports ITU V.152 - Procedures for supporting Voice-Band Data over IP Networks.

Remote party ID <draft-ietf-sip-privacy-04.txt>.

Supports obtaining Proxy Domain Name(s) from DHCP (Dynamic Host Control Protocol) according to RFC 3361.

Supports handling forking proxy multiple responses.

RFC 2833 Relay for DTMF Digits, including payload type negotiation.

DTMF out-of-band transfer using:

• INFO method <draft-choudhuri-sip-info-digit-00.txt>

• INFO method, compatible with Cisco gateways

• NOTIFY method <draft-mahy-sipping-signaled-digits-01.txt>

• INFO method, compatible with Korea Telecom format

SIP URL: sip:”phone number”@IP address (such as [email protected], where “122556” is the phone number of the source or destination) or sip:”phone_number”@”domain name”, such as [email protected]. Note that the SIP URI host name can be configured differently per called number.

Supports RFC 4040, RTP payload format for a 64 kbit/s transparent data.

Can negotiate coder from a list of given coders.

Supports negotiation of dynamic payload types.

Supports multiple ptime values per coder.

Supports RFC 3389, RTP Payload for Comfort Noise.

Supports RFC 3824, Using E.164 numbers with SIP (ENUM).

Supports receipt and DNS resolution of FQDNs received in SDP.

Supports <draft-ietf-sip-gruu-09>, Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in SIP

Responds to OPTIONS messages both outside a SIP dialog and in mid-call. Generates SIP OPTIONS messages as Proxy keep-alive mechanism.

Publishes the total number of free Tel channels in a 200 OK response to an OPTIONS requests.

Support RFC 3310, HTTP Digest Authentication Using Authentication and Key Agreement (AKA).

Supports 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 3rd-Generation Partnership Project (3GPP) [Partial].


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

User Agent Client (UAC) Yes

User Agent Server (UAS) Yes

Proxy Server Third-party only tested with, amongst others, Ubiquity, Delta3, Microsoft, 3Com, BroadSoft, Snom and Cisco Proxies)

Redirect Server Third-party

Registrar Server Third-party

Event Publication Agent (EPA)

Yes

Event State Compositor (ESC)

Third-party

2.2.2 SIP Methods The device supports the following SIP Methods:

Table 2-2: Supported SIP Methods

Method Supported Comments

INVITE Yes

ACK Yes

BYE Yes



Method Supported Comments

CANCEL Yes

REGISTER Yes Send only

REFER Yes Inside and outside of a dialog

NOTIFY Yes

INFO Yes

OPTIONS Yes

PRACK Yes

UPDATE Yes

PUBLISH Yes Send only

SUBSCRIBE Yes

2.2.3 SIP Headers The device supports the following SIP Headers:

Table 2-3: Supported SIP Headers

Header Field Supported

Accept Yes

Accept–Encoding Yes

Alert-Info Yes

Allow Yes

Also Yes

Asserted-Identity Yes

Authorization Yes

Call-ID Yes

Call-Info Yes

Contact Yes

Content-Disposition Yes

Content-Encoding Yes

Content-Length Yes

Content-Type Yes

Cseq Yes

Date Yes

Diversion Yes

Encryption No

Expires Yes


MediaPack Series


Fax Yes

From Yes

History-Info Yes

Join Yes

Max-Forwards Yes

Messages-Waiting Yes

MIN-SE Yes

Organization No

P-Associated-URI Yes (Receive Only)

P-Asserted-Identity Yes

P-Charging-Vector Yes

P-Preferred-Identity Yes

Priority Yes

Proxy- Authenticate Yes

Proxy- Authorization Yes

Proxy- Require Yes

Prack Yes

Reason Yes

Record- Route Yes

Refer-To Yes

Referred-By Yes

Replaces Yes

Require Yes

Remote-Party-ID Yes

Response- Key Yes

Retry-After Yes

Route Yes

Rseq Yes

Session-Expires Yes

Server Yes

Service-Route Yes

SIP-If-Match Yes

Subject Yes

Supported Yes

Target-Dialog Yes

Timestamp Yes




To Yes

Unsupported Yes

User- Agent Yes

Via Yes

Voicemail Yes

Warning Yes

WWW- Authenticate Yes

2.2.4 SDP Headers The device supports the following SDP Headers:

Table 2-4: Supported SDP Headers

SDP Header Element Supported

v - Protocol version Yes

o - Owner/ creator and session identifier Yes

a - Attribute information Yes

c - Connection information Yes

d - Digit Yes

m - Media name and transport address Yes

s - Session information Yes

t - Time alive header Yes

b - Bandwidth header Yes

u - Uri Description Header Yes

e - Email Address header Yes

i - Session Info Header Yes

p - Phone number header Yes

y - Year Yes

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


MediaPack Series

2.2.5.1 1xx Response – Information Responses

Table 2-5: Supported 1xx SIP Responses

1xx Response Supported Comments

100 Trying Yes The SIP device generates this response upon receiving a Proceeding message from ISDN or immediately after placing a call for CAS signaling.

180 Ringing Yes The SIP device generates this response for an incoming INVITE message. Upon receiving this response, the device waits for a 200 OK response.

181 Call is Being Forwarded

Yes The 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.

182 Queued Yes 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.

183 Session Progress

Yes The SIP device generates this response if the Early Media feature is enabled and if the device plays a Ringback tone to IP

2.2.5.2 2xx Response – Successful Responses



200 OK Yes --

202 Accepted Yes --

2.2.5.3 3xx Response – Redirection Responses



300 Multiple Choice Yes The device responds with an ACK, and then resends the request to the first new address in the contact list.

301 Moved Permanently

Yes The device responds with an ACK, and then resends the request to the new address.

302 Moved Temporarily

Yes The 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.

305 Use Proxy Yes The device responds with an ACK, and then resends the request to a new address.

380 Alternate Service

Yes The device responds with an ACK, and then resends the request to a new address.



2.2.5.4 4xx Response – Client Failure Responses



400 Bad Request Yes The device doesn't generate this response. Upon receipt of this message, and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

401 Unauthorized Yes Authentication support for Basic and Digest. Upon receiving this message, the device issues a new request according to the scheme received on this response.

402 Payment Required

Yes The device doesn't generate this response. Upon receipt of this message, and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

403 Forbidden Yes The device doesn't generate this response. Upon receipt of this message, and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

404 Not Found Yes The 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.

405 Method Not Allowed

Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

406 Not Acceptable Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

407 Proxy Authentication

Required

Yes Authentication support for Basic and Digest. Upon receiving this message, the device issues a new request according to the scheme received on this response.

408 Request Timeout Yes The device generates this response if the no-answer timer expires. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

409 Conflict Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

410 Gone Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

411 Length Required Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

413 Request Entity Too Large


415 Unsupported Media

Yes If the device receives a 415 Unsupported Media response, it notifies the User with a Reorder Tone. The device generates this response in case of SDP mismatch.


MediaPack Series


420 Bad Extension Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

423 Interval Too Brief Yes The device does not generate this response. On reception of this message the device uses the value received in the Min-Expires header as the registration time.

433 Anonymity Disallowed

Yes If the device receives a 433 Anonymity Disallowed, it sends a DISCONNECT message to the PSTN with a cause value of 21 (Call Rejected). In addition, the device can be configured, using the Release Reason Mapping, to generate a 433 response when any cause is received from the PSTN side.

480 Temporarily Unavailable

Yes If the device receives a 480 Temporarily Unavailable response, it notifies the User with a Reorder Tone. This response is issued if there is no response from remote.

481 Call Leg/Transaction Does Not Exist


482 Loop Detected Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

483 Too Many Hops Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

484 Address Incomplete


485 Ambiguous Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

486 Busy Here Yes The SIP device generates this response if the called party is off-hook and the call cannot be presented as a call waiting call. Upon receipt of this response, the device notifies the User and generates a busy tone.

487 Request Canceled

Yes This response indicates that the initial request is terminated with a BYE or CANCEL request.

488 Not Acceptable Yes The device doesn't generate this response. Upon receipt of this message and before a 200 OK has been received, the device responds with an ACK and disconnects the call.

491 Request Pending Yes When acting as a UAS: the device sent a re-INVITE on an established session and is still in progress. If it receives a re-INVITE on the same dialog, it returns a 491 response to the received INVITE.

When acting as a UAC: If the device receives a 491 response to a re-INVITE, it starts a timer. After the timer expires, the UAC tries to send the re-INVITE again.



2.2.5.5 5xx Response – Server Failure Responses


5xx Response Comments

500 Internal Server Error

501 Not Implemented

502 Bad gateway

503 Service Unavailable

504 Gateway Timeout

505 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


6xx Response Comments

600 Busy Everywhere

603 Decline

604 Does Not Exist Anywhere

606 Not Acceptable

Upon receipt of any of these Responses, the device releases the call, sending an appropriate release cause to the PSTN side.


MediaPack Series

Reader's Notes


SIP Release Notes 3. Known Constraints

3 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.

2. The number of RTP payloads packed in a single G.729 packet (M channel parameter) is limited to 5.

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.

2. Debug Recording:

• Only one IP target is allowed.

• Maximum of 50 trace rules are allowed simultaneously.

• Maximum of 5 media stream recordings are allowed simultaneously.

3. VLAN Pass-Through mode is not supported.

4. 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.

5. 10Base-T Half-Duplex is not supported (only 10/100Base-T Full Duplex and 100Base-T Half-Duplex are supported).

6. 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 100Base-TX instead of 10Base-T (due to the larger bandwidth).

7. ThroughPacket RTP Multiplexing is not supported with VLAN configurations.


MediaPack Series

8. PPPoE is not supported.

9. 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 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 can’t 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.

2. 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.

3. The following Web interface's configuration pages do not support the Scenario mode:

• Web User Accounts

• Web & Telnet Access List

• Regional Settings

4. 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.

5. The Routing table can be configured in the Web interface, but the ini file is not updated with the new settings.

6. This version does not support screen resolution 1152 x 864.

7. On the 'IP Settings' page, when selecting a 'multiple' or 'dual' value from the 'IP Networking Mode' field, the 'DHCP' field is incorrectly enabled.

8. Not all parameters can be changed on-the-fly in the Web interface. Parameters that can't be changed on-the-fly are noted with the lightning symbol. To change these parameters, reset the device using the Web interface's Reset button.

9. 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.

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.


MediaPack Series

3.4 SNMP Constraints This release includes the following known SNMP constraints:

1. In the ipCidrRouteTable, new rows cannot be added and rows that were previously deleted using the Web interface, cannot be deleted.

2. When configuring the acSysInterfaceTable using SNMP or the Web interface, validation is only performed after device reset.

3. When attempting to enable Telnet using SNMP, a fail notification is displayed despite the operation being successful.

4. 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.

5. The SNMPv3 users table returns the “line removed“ notice when adding a new row to an active row index.

6. After adding an empty line to the SNMPV3 table, it's impossible to delete it or add new lines.

7. The default values created in an IPSec configuration table are incorrect. The user should override the default values before activating the new row.

8. The acBoardConfigurationError alarm trap, generated as a result of a configuration error, does not clear.

9. The following RTP MIB objects are not supported: rtpRcvrSRCSSRC, rtpRcvrSSRC, rtpSenderSSRC, rtpRcvrLostPackets, rtpRcvrPackets, rtpSenderPackets, rtpRcvrOctets, and rtpSenderOctets.

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 4. Resolved Constraints

4 Resolved Constraints The following constraints from previous releases have been resolved in Release 5.4:

1. Web Interface: When configuring the RadiusAuthServerIp parameter with a non-existent server IP, the BehaviorUponRadiusTimeout parameter value is ignored.

2. Web Interface: Users cannot enter a username/password of 8 characters.


MediaPack Series

Reader's Notes


SIP Release Notes 5. Earlier Releases

5 Earlier Releases Details of previous releases can be found in the Release Notes of Version 5.2, published by AudioCodes on May-24-2007.

Release Notes

Version 5.4

www.audiocodes.com

http://www.audiocodes.com/

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

Documents

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