Mediant 2000™ SIP Gateway & TP-1610/SIP cPCI Board

Mediant 2000™ SIP Gateway & TP-1610/SIP cPCI Board

Release Notes for Preliminary Version

4.2 Beta

Document #: LTRT -00690

US : AudioCodes Inc, 2890 Zanker Road, Suite # 200, San Jose, CA 95134.

Tel: 408-577-0488 - Fax: 408-577-0492

Internationa l: AudioCodes Ltd, 4 HaHoresh Street, PO Box 14, Yehud 56470, Israel.

Tel: +972-3-539 4000 - Fax: +972-3-539 4040

China: AudioCodes Ltd, Room 209 Tower A, Grand Pacific Building,

8A Guanghua Road, Chaoyang District, Beijing 100026, P.R.China

Tel: 86-10-6581-6622 - Fax: 86-10-6581-5599

Technical Support: [email protected]

Mediant 2000 Gateway SIP Release Notes

Software Version 4.2 Beta 3 May 2003

Notice

This document describes the release of the AudioCodes Mediant 2000/SIP Gateway, and TP-1610 /SIP cPCI board, Version 4.2 Beta, supporting IETF SIP (Session Initialization Protocol).

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 the accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions.

For Technical Support please contact: e-mail: [email protected]

In the US: fax 408-577-0492 In other countries: fax +972-3-539-4040

© 2003 AudioCodes Ltd. All rights reserved. This document is subject to change without notice.

Date Published: Feb-17-2003 Date Printed: May-28-2003

Acknowledgements

AC, AudioCoded, AudioCodes, AudioCodes logo, IPmedia, Mediant, MediaPack, NetCoder, Stretto, TrunkPack and VoicePacketizer, are trademarks or registered trademarks of AudioCodes Limited.

All other products or trademarks are property of their respective owners.

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.

Mediant 2000 Gateway

SIP Release Notes 4 Document #: LTRT-00690

Documentation

For a detailed description of the Mediant 2000/SIP Gateway and the TP-1610/SIP cPCI Board, refer to the AudioCodes Mediant 2000/SIP Gateway User’s Manual, Document #: LTRT-00688.

Mediant 2000 Gateway SIP Release Notes


Table of Contents

1. What’s New in this Release .............................................................................. 7 1.1 New SIP Features ......................................................................................... 7 1.2 Other New Features ..................................................................................... 7 1.3 New Parameters in ini file .............................................................................10 1.4 Resolved Constraints....................................................................................17

2. SIP and PSTN Compatibility ............................................................................19 2.1 PSTN to SIP Interworking .............................................................................19 2.2 Supported SIP Features ...............................................................................20 2.3 Unsupported SIP Features ............................................................................21 2.4 SIP Compliance Tables ................................................................................22

2.4.1 SIP Functions .........................................................................................22 2.4.2 SIP Methods...........................................................................................23 2.4.3 SIP Headers ...........................................................................................23 2.4.4 SDP Headers ................................ ................................ ..........................25 2.4.5 SIP Responses.......................................................................................26

2.4.5.1 1xx Response – Information Responses ............................................................ 26 2.4.5.2 2xx Response – Successful Responses.............................................................. 27 2.4.5.3 3xx Response – Redirection Responses ............................................................ 27 2.4.5.4 4xx Response – Request Failure Responses.................................................... 28 2.4.5.5 5xx Response – Server Failure Responses........................................................ 32 2.4.5.6 6xx Response – Global Responses..................................................................... 32

3. Functional Constraints .....................................................................................33 3.1 Mediant 2000 Gateway Clock Settings...........................................................35

4. Mediant 2000 SIP Kit........................................................................................36 5. Version History.................................................................................................38

5.1 Version 4.01................................................................................................38 5.1.1 New Features .........................................................................................38 5.1.2 Version 4.01 New Parameters ................................ ................................ ..39 5.1.3 Resolved Bugs ........................................................................................41

5.2 Version 4.0 GA ............................................................................................42 5.2.1 New Features .........................................................................................42 5.2.2 Resolved Bugs ........................................................................................43

5.3 Version 4.0 Beta................................ ................................ ..........................43



List of Tables Table 1-1: Description of ini File Parameters (continued on pages 8 to 10)...................... 10 Table 2-1: Supported SIP Functions ............................................................................ 22 Table 2-2: Support ed SIP Methods.............................................................................. 23 Table 2-3: Supported SIP Headers (continues on pages 16 to 24) .................................. 23 Table 2-4: Supported SDP Headers................................ ................................ ............. 25 Table 2-5: Supported 1xx SIP Responses .................................................................... 26 Table 2-6: Supported 2xx SIP Responses .................................................................... 27 Table 2-7: Supported 3xx SIP Responses .................................................................... 27 Table 2-8: Supported 4xx SIP Responses (continues on pages 20 to 23) ........................ 28 Table 2-9: Supported 5xx SIP Responses .................................................................... 32 Table 2-10: Supported 6xx SIP Responses .................................................................. 32

Mediant 2000 Gateway What’s New in this Release


1. What’s New in this Release 1.1 New SIP Features

• Redirect Proxy server support (sending subsequent INVITE requests to the destination(s) indicated in SIP contact header(s).

• Redundant Proxy support (similar to GK redundancy) and fallback to internal routing table if Proxy is no responding.

• Implementation of new Offer/Answer SDP negotiation as per RFC 3264

• SIP session timer support.

• Support blind and consultation call transfers,by using REFER, Referred-By and Replaces .

• Out of band DTMF support using INFO messages

• Support of “tel” and "fax" URL formats and accepting numbers such as sip:[email protected].

• Support receive only mode IP address=0.0.0.0 in “c” field

• PRACK reliable provisional responses <rfc 3262>

• Diversion Header <draft-levy-sip-diversion-05.txt>

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

1.2 Other New Features • IP Security feature. Using "SecureCallsFromIP = 1" definition, it is possible

to define list of IP addresses from where the gateway accepts calls. Other calls are rejected. The list of IP addresses is taken from destination phone prefix table.



• Optional Playback of ringback tones to IP and to PSTN and Using the “Progress Indicator IE” for decision of playing or not playing the ring back tone (to PSTN and to IP)

• Support of additional coders: G.726/32Kbps, G.723/5.3, G.729 10msec

• Trunk group support. New "IP to Trunk group" routing table defines routing rules per called number prefix to one of the trunk groups

• On-the-fly (from Web) configuration of gateway phone numbers and trunk groups

• Increasing Phone to IP routing table capacity up to 50 IP addresses.

• Differentiated Services parameter (for RTP packets) was added in addition to previous Type of service and IP preceding parameters.

• Online trunk configuration from Web browser is now supported. The user can now configure new trunks and change the configuration of existing trunks without restarting the gateway. Any trunk can be configured to be out of service (by sending an AIS alarm).

• Busy-out feature. If no LAN or No Proxy available all E1/T1 trunks will be automatically put out of service by sending a remote alarm (such as AIS) or Service disconnection message for T1 PRI trunks.

• B channel negotiation, supporting exclusive or preferred modes .

• Support of ISDN (only T1 trunks) non Facility Associated signaling (NFAS) .

• Using the “Progress Indication IE” for decision of playing or not playing the ring back tone (to PSTN and to IP).

• Interworking of Q.931/PRI Display element into SIP From header.

• Option, to swap between DNIS and Redirect numbers for ISDN incoming call.

• Source Number Manipulation (same as Destination number manipulation) can be used for both Tel to IP and IP to Tel directions.

• Number manipulation tables are now supported on-the-fly from Web browser.



• Progress Indicator parameter (for PRI calls) can be configured from Web/INI file

• Different modes of Play or not to play Ringback tone to Tel and to IP can be configured on-the-fly from Web.

• Support for a special Jitter Buffer optimization mode for Modem or Fax. User should set the jitter buffer Optimization Factor to 13 for Modem or Fax calls.

• "Save Configuration" button was added to enable permanently storing of gateway configuration parameters to flash memory, without resetting the gateway.

• INI file can now be uploaded from the gateway to PC, controllable from Web Browser. The INI file will contain all configuration parameters that are different from their default value.

• HTTP port for Web management can be configured via INI file, using "HTTPport" parameter

• System date and Time can be configured from Web browser and are shown in "Versions" Web page.

• LAN watchdog. If enabled, the gateway will be restarted if internal LAN failure is detected.

• Implementation of “Audit” task, to monitor and to release “stuck” resources and to overcome overload gateway conditions.

• Syslog CDR (Call Detail Record) is generated at each call end, reporting call session data to Syslog server.



1.3 New Parameters in ini file Note: Most of the parameters below can be configured from both ini file and by using Web browser.

Table 1-1: Description of ini File Parameters (continued on pages 10 to 16)

ini File [Web] Parameter Name

Description

Individual Trunk Setting Parameters

FramingMethod_x

[Framing Method]

Same as "FramingMethod" for specific Trunk ID (x=0 to 7)

ProtocolType_x

[Protocol Type]

Same as "ProtocolType" for specific Trunk ID (x=0 to 7)

TerminationSide_x

[ISDN Termination Side]

Same as "TerminationSide" for specific Trunk ID (x=0 to 7)

ClockMaster_x [Clock Master]

Same as "ClockMaster" for specific Trunk ID (x=0 to 7)

LineCode_x

[Line Code]

Same as "LineCode" for specific Trunk ID (x=0 to 7)

ISDNIBehavior_x

[Q931 Layer Response Behavior]

Same as "ISDNBehavior" for specific Trunk ID (x=0 to 7)

ISDNInCallsBehavior_x

[Incoming Calls Behavior]

Same as "ISDNInCallsBehavior" for specific Trunk ID (x=0 to 7)

NFASGroupNumber_x

[NFAS Group Number]

0 – non NFAS trunk (default)

1 to 4 - NFAS group number

Indicates the NFAS group number for the selected trunk.

"x" idenfines the Trunk ID (0-7).

The trunks that belongs to same NFAS group will have the same number.

With ISDN Non-Facility Associated Signalling you can use single D-channel to control multiple PRI interfaces.

Applicable only for T1 ISDN protocols.

The "NFAS member" of each trunk in NFAS group is equal to the Trunk ID (0-7).





Description

DchConfig_x

[D-Channel Configuration]

0 – Primary Trunk (default)

1 – Backup Trunk 2 – NFAS Trunk

D-channel configuration parameter defines primary, backup (optional) and B -channels only trunks. "x" identifies the Trunk ID (0-7).

Primary trunk contains D -channel that is used for signaling.

Backup trunk contains backup D-channel, that will be used if the primary D-channel fails.

The other NFAS trunks, contain only 24 B-channels, without signaling D-channel.

Applicable only for T1 ISDN protocols Backup trunk is not supported for DMS PRI variants

CASTableIndex_x 0-7

The parameter defines CAS protocol per Trunk ID (x=0 to 7).

The CAS protocol index value is according to the "Y" value in "CASFileName_Y"

CASFileName_0

CasFileName_ 1

;

CASFileName_7

CAS file name (such as "E_M_WinkTable.dat") defines the CAS protocol. It is possible to download up to 8 different CAS files by repeating the CASFileName parameter, Each CAS file can be associated with one or more of the gateway trunks using "CASTableIndex_x" parameter.

CASTableNum 1 – 8

Indicates how many CAS protocol configurations files are loaded.

IPDiffServ [RTP IP Diff Serv

(in Network Setting page) ]

0 – 63 value for setting the Diff Services Code Point (DSCP). If defined it will be used instead of IP TOS and IP Precedence settings.

Applies only to RTP packets

StaticNATIP Static NAT IP address.





Description

[NAT IP address] Global gateway IP address. Should be defined if static Network Address Translation device (NAT) is used between the gateway and Internet.

ProxyIP

ProxyIP

[Proxy IP address]

[Redundant Proxy IP address]

IP addresses of main and redundant sip proxies. The gateway will use redundant IP address if main proxy is not responding. After switching to redundant Proxy, the gateway will continue working with it until next failure. The gateway will monitor connection with working proxy using keep-alive messages ("OPTIONS").

BchannelNegotiation

[B-channel negotiation]

0 – Exclusive (default)

1 – Preferred

2 - Any

EnableBusyOut [PSTN Busy Out]

0 – Not used 1 – Enable busy out

If no LAN or No Proxy available all E1/T1 trunks will be automatically put out of service by sending a remote alarm (such as AIS) or Service disconnection message for T1 PRI trunks.

TrunkGroup_x

[Trunk Group table]

T/a-b,c

x – Trunk group ID, starting with 1 [1 to 99]

T – Trunk physical port (1 to 8) a – Starting B-channel (from 1)

b – Ending B-channel (up to 31)

c – phone number allocated with the first channel

For example: TrunkGroup_1 = 1/1-31,1000 (for E1 span)

TrunkGroup_1 = 2/1-31,2000

TrunkGroup_2 = 3/1-24,3000 (for T1 span)

Trunk group is an optional parameter, it is possible still to use "ChannelList" parameter to define enabled B-channels as single group.





Description

PSTNPrefix

[IP to Trunk group routing table]

a,b

a – destination number prefix (for IPàTel calls)

b – Trunk group ID

Outgoing IPàTel calls starting with "a" will be routed to Trunk group "b"

NumberMapTel2IP

[Manipulation Tables:

TEL à IP Destination Numbers]

a,b,c,d a = Phone number prefix b = Number of stripped digits c = New prefix to be added, or "none" d = Number of remained digits from right

The parameter defines rule(s) for changing the destination phone number, received in Telà IP incoming call. Each rule is applied to a number that matches the "a" prefix. The "b" and "c" actions apply to a number that starts with "a" prefix. The actions apply to the number in the following order: "b", "d" and then "c".

NumberMapIP2Tel


IP à TEL Destination Numbers]

a,b,c,d,e,f a = Phone number prefix b = Number of stripped digits c = New prefix to be added, or "none"

d = Number of remained digits from right

e = H.225/Q.931 Number Plan

f = H.225/Q.931 Number Type

The parameter defines rule(s) for changing the destination phone number, received in IPàTel outgoing call. Each rule is applied to a number that matches the "a" prefix. The "b" and "c" actions apply to a number that starts with "a" prefix. The actions apply to the number in the following order: "b", "d" and then "c".

Parameters can be skipped by "$$", such as NumberMapIP2TEL=01,2,972,$$,0,0

SourceNumberMapTEL2IP Same rules as for destination number





Description


TEL à IP Source Numbers]

SourceNumberMapIP2TEL


IP à TEL Source Numbers]

Same rules as for destination number

AddTrunkGroupAsPrefix

[Add Trunk Group ID as Prefix]

0 – not used

1 - For TelàIP incoming call, Trunk Group ID will be added as prefix to destination phone number. Applicable only if Hunt group ID are configured.

Can be used to define various routing rules.

RouteModeIP2Tel

[IP to TEL routing mode]

0 – Route calls before number manipulation (default)

1 – Route calls after number manipulation Defines order between routing calls to Trunk group and manipulation of destination number

RouteModeTel2IP

[TEL to IP routing mode]

0 – Route calls before number manipulation (default)

1 – Route calls after number manipulation

Defines order between routing incoming calls to IP, using routing table, and manipulation of destination number

Not applicable if PROXY routing is used.

PlayRBTone2Tel

[Play Ringback Tone to TEL]

0 – Don't play

1 – Play

2 – Play according to 180/183 response.

If 183 session progress with SDP message was received, the GW will cut through the voice channel and it will not play ringback tone. PI=8 will be send in ISDN alert message (unless "Progress Indicator to ISDN" parameter is configured differently.

PlayRBTone2IP

[Play Ringback Tone to IP]

0 – Don't play (default)

1 – The GW will play ringback tone to IP after sending SIP 183 session progress response.

ProgressIndicator2ISDN

[Progress Indicator to ISDN]

0,1 or 8

If set to "0" no PI will be send to ISDN

If set to "1" or "8" the PI value will be send to PSTN in





Description

Q.931/Proceeding and Alerting messages.

If not configured (default), the PI in ISDN messages will be set according to "Play Ringback to Tel" parameter.

Usually if PI=1or 8, the PSTN/PBX switch will cut through the audio channel without playing local ringback tone, enabling originating party to hear remote call progress tones or network announcements

SecureCallsFromIP 0 = Accept all SIP Calls (default) 1 = Accept SIP calls only from IP addresses defined in Prefix routing table. When SecureCalsFromIP is enabled, gateway rejects calls from unknown IP addresses. When working with Proxy, you can set the Proxy IP address in the Prefix routing table, to accept calls only from the Proxy, and reject other calls.

EnableTransfer

[Enable Transfer]

0 – Call t ransfer is not allowed

1 - The gateway will perform call transfer if it receives a REFER message

SwapReddirectNumber [Swap Redirect and Called Numbers]

0 - Don't change numbers (default) 1- Swap between DNIS and Redirect numbers for ISDN incoming call

IsPrackRequired

[Enable PRACK]

0 – PRACK is not used

1 – PRACK will be used by gateway

SipSessionExpires

[Session-Expires Time]

1 – 100 [sec]

Timeout for Keep alive "re-INVITE" message within a SIP session

EnableProxyKeepAlive

[Enable Proxy Keep Alive]

0 – Disable (default)

1 – Keep alive with Proxy, by sending "OPTIONS" sip message every 60 seconds

EnableRPIheader 0 – Disable 1 - RPI header will be generated in sip INVITE message for called, calling and redirect numbers.





Description

RemovePrefix 0 – Don't remove

1 – remove PSTN prefix (from routing table) before forwarding the called phone number to PSTN (default)

Applicable only if number manipulation feature is configured to be used after call routing, for IP to PSTN calls.

IsDTMFUsed [Use INFO for DTMF]

0 – not used 1- INFO message is used to transfer DTMF digits using

IETF draft "draft-choudhuri-sip-info-digit-00".

To use out of band DTMF transfer you should disable inband DTMF by setting:

"DTMFTransportType = 0" (erase digit and don't relay)

EnableCDR [Enable Syslog CDR]

0 – CDR is not used 1- Call detail record will be send to Syslog server at each call end.

The CDR Syslog message complies to rfc 3161 and is identified by:

Facility = 17 (local1) and Severity = 6 (Informational)

EnableLANWatchDog 0 – Disable LAN WatchDog (default) 1 – Enable LAN WatchDog

The gateway will be restarted if internal LAN failure is detected

HTTPport HTTP port used for Web management (default = 80)



1.4 Resolved Constraints • Changing of IP parameters, including IP address, Subnet and default gateway

can be done on the fly, without a need for gateway reset.

• Support for reordered RTP packets – a new algorithm has been implemented to handle RTP packets which are received reordered. This feature improves the voice quality on a network which suffers from packet reordering problems.

• Support of different CAS and ISDN protocol variants per each gateway E1/T1 span.

• The gateway can support calls between E1/T1 spans (on the same gateway module) with or without a Proxy.

• Received PSTN calls with same calling and called numbers are now handled OK.



Reader's Notes

Mediant 2000 Gateway SIP and PSTN Compatibility


2. SIP and PSTN Compatibility 2.1 PSTN to SIP Interworking

Mediant 2000/SIP Gateway supports various ISDN PRI protocols such as EuroISDN, North American NI2, Lucent 5ESS, Nortel DMS100, Japan J1, as well as QSIG (basic call). PRI support includes User Termination or Network Termination side. ISDN-PRI protocols can be defined on an E1/T1 basis (i.e., different variants of PRI are allowed on different E1/T1 spans).

In addition, it supports numerous variants of CAS protocols for E1 and T1 spans, including MFCR2, E&M wink start, E&M immediate start, E&M delay dial/start, loop-start, and ground start. CAS protocols can be defined on an E1/T1 basis (i.e., different variants of CAS are allowed on different E1/T1 spans).

PSTN to SIP and SIP to PSTN Called number can be optionally modified according to rules that are defined in gateway ini file.

The following features are supported:

Ø Definition and using of Trunk Groups for routing IPà PSTN calls.

Ø B-channel negotiation for PRI spans .

Ø ISDN Non Facility Associated Signaling (NFAS).

Ø PRI to SIP Interworking of Q.931 Display (Calling name) information element.

Ø Configuration of Numbering Plan and Type for IPà ISDN calls

Ø Interworking of PSTN to SIP release causes

Ø Interworking of ISDN to diversion number (as per IETF draft-levy-sip-diversion-05.txt)

The following features aren’t supported:

Ø Q.931 and QSIG supplementary services.



Ø Overlap sending & receiving (only en-bloc sending is used).

Ø QSIG Calling name identification

Ø QSIG and PRI connected line identification

2.2 Supported SIP Features The Mediant 2000/SIP main features are:

• Works with Proxy or without Proxy, using an internal routing table.

• Fallback to internal routing table if Proxy is not responding

• Supporting redundant Proxies (up to three proxy IP addresses can be configured). If main Proxy fails redundant proxy will be used.

• Proxy Registration, such as:

REGISTER sip:proxyname SIP/2.0 VIA: SIP/2.0/UDP 212.179.22.229;branch=z9hG4bRaC7AU234 From: <sip:Username@sipgatewayname>;tag=1c29347 To: <sip:Username@sipgatewayname> Call-ID: [email protected] Seq: 1 REGISTER Expires: 3600 Contact: sip:[email protected] Content-Length: 0

Where the “proxyname”, “Username” and “sipgatewayname” strings are defined in the Mediant ini file (or configured from the Web).

• Proxy and Registrar Authentication (handling 401 and 407 responses) using Basic or Digest methods.

• Supported methods: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS , INFO and REFER .

• Modifying connection parameters in a call (re-INVITE)

• Working with Redirect server and handling 3xx responses



• Early media (supporting 183 Session Progress).

• PRACK reliable provisional responses <rfc 3262>

• Diversion Header <draft-levy-sip-diversion-05.txt>

• Call Transfer Supplem entary services using REFER, Refer-To, Referred-By, Replaces and NOTIFY .

• Session Timer <draft-ietf-sip-session-timer-10.txt>

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

• DTMF Payload for DTMF Digits <rfc 2833>

• DTMF out of band transfer using INFO method

• SIP URL: sip:”phone number”@IP address (such as [email protected], where “122” is the phone number of the source or destination phone number) or sip:”phone_number”@”domain name”, such as [email protected]

• Reliable UDP transport, with retransmissions

• Supported codecs: G.711 A-law, G.711 µ-law, G.723 (6.3 kbps), G.729A , NetCoder (6.4 to 8.8 kbps).

• Can negotiate codec from a list of given codecs.

2.3 Unsupported SIP Features The following SIP features are not supported:

• NOTIFY, SUBSCRIBE and UPDATE methods

• Tel URI (Uniform Resource Identifier) as described in RFC 2806bis

• Supplementary services (MWI,, call forwarding,..)

• Network Asserted Identity (rfc 3325)

• T.38 real time FAX (using SIP)



• Multicast

• TCP/IP transport layer and TLS

2.4 SIP Compliance Tables The MEDIANT 2000/SIP (Session Initialization Protocol) Gateway complies with RFC 3261, as shown in the following sections.

2.4.1 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 (Checked with Ubiquity, Delta3, Microsoft, 3Com, Snom and Cisco Proxies)

Redirect Server Third-party

Registrar Server Third -party



2.4.2 SIP Methods

Table 2-2: Supported SIP Methods

Method Supported Comments

INVITE Yes

ACK Yes

BYE Yes

CANCEL Yes

REGISTER Yes Send only

REFER Yes

NOTIFY Yes

INFO Yes

OPTIONS Yes

PRACK Yes

UPDATE No

2.4.3 SIP Headers Release 4.01 of the Mediant 2000/SIP Gateway supports the following SIP Headers:

Table 2-3: Supported SIP Headers (continues on pages 23 to 32)

Header Field Supported

Accept Yes

Accept–Encoding Yes

Alert-Info Yes

Allow Yes

Also Yes

Authorization Yes

Call-ID Yes





Call-Info Yes

Contact Yes

Content-Encoding Yes

Content-Length Yes

Content-Type Yes

Cseq Yes

Date Yes

Encryption No

Expires Yes

Fax Yes

From Yes

Max -Forwards Yes

Messages-Waiting Yes

MIN-SE Yes

Organization No

Priority No

Proxy- Authenticate Yes

Proxy- Authorization Yes

Proxy- Require Yes

Prack 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

Subject Yes

Supported Yes

Timestamp Yes

To Yes

Unsupported Yes

User- Agent Yes

Via Yes

Voicemail Yes

Warning Yes

WWW- Authenticate Yes

2.4.4 SDP Headers Release 4.01 of the Mediant 2000/SIP Gateway 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

2.4.5 SIP Responses Release 4.0.1 of the Mediant 2000/SIP Gateway supports the following SIP responses:

1xx Response - Information Responses.

2xx Response - Successful Responses.

3xx Response - Redirection Responses.

4xx Response - Request Failure Responses.

5xx Response - Server Failure Responses.

6xx Response - Global Responses.

2.4.5.1 1xx Response – Information Responses

Table 2-5: Supported 1xx SIP Responses

1xx Response Supported Comments

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

180 Ringing Yes The SIP Gateway generates this response for an incoming INVITE





message. On receiving this response, the Gateway waits for a 200 OK response.

181 Call is being forwarded

Yes The SIP Gateway does not generate these responses. However, the Gateway does receive them. The Gateway processes these responses the same way that it processes the 100 Trying response.

182 Queued Not

183 Session Progress Yes The SIP Gateway will generate this response if Early Media feature is enabled

2.4.5.2 2xx Response – Successful Responses



200 OK Yes

202 Accepted Yes

2.4.5.3 3xx Response – Redirection Responses





300 Multiple Choice

Yes The Gateway responds with an Ack and resends the request to first in the contact list, new address.

301 Moved Permanently

Yes The Gateway responds with an Ack and resends the request to new address.

302 Moved Temporarily

Yes "

305 Use Proxy Yes "

380 Alternate Service

Yes "

2.4.5.4 4xx Response – Request Failure Responses

Table 2-8: Supported 4xx SIP Responses (continues on pages 28 to 31)


400 Bad Request Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an Ack and disconnects the call.

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

402 Payment Required

Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.

403 Forbidden Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.

404 Not Found Yes The SIP Gateway generates this response if it is unable to locate the callee. On receiving this response, the Gateway notifies the User with a Reorder Tone.





405 Method Not Allowed


406 Not Acceptable Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.

407 Proxy Authentication Required

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

408 Request Timeout Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.

409 Conflict Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.

410 Gone Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.

411 Length Required Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.

413 Request Entity Too Large


414 Request-URL Too Long






415 Unsupported Media

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

420 Bad Extension Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.

480 Temporarily Unavailable

Yes If the Gateway 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 Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.

483 Too Many Hops Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.

484 Address Incomplete


485 Ambiguous Yes The Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.





486 Busy Here Yes The SIP Gateway generates this response if the called party is off hook and the call cannot be presented as a call waiting call. On receiving this response, the Gateway 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 Gateway does not generate this response. On reception of this message, before a 200OK has been received, the gateway responds with an ACK and disconnects the call.



2.4.5.5 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

On reception of any of these Responses, the GW will release the call, sending appropriate release cause to PSTN side.

The GW generates 5xx response according to PSTN release cause coming from PSTN.

2.4.5.6 6xx Response – Global Responses


6XX Response Comments

600 Busy Everywhere

603 Decline

604 Does Not Exist Anywhere

606 Not Acceptable

On reception of any of these Responses, the GW will release the call, sending appropriate release cause to PSTN side.

Mediant 2000 Gateway Functional Constraints


3. Functional Constraints • The Mediant 2000 gateway comprises of two independent gateway modules,

GW-1 and GW-2 each with its own MAC and IP addresses.

• It is strongly recommended to operate the gateway with 100 Base-T switches. Use of 10 Base-T LAN hubs should be avoided.

• When updating software version (from Web or TFTP) both ram.cmp and ini files should be loaded.

• The Mediant 2000 Gateway supports only symmetrical codecs.

• Voice gain, silence suppression, Jitter buffer and other channel parameters are preset in the ini file as a single value for all channels.

• When using G.711 with 20 msec codecs packetization time (without silence suppression) the Mediant gateway can support up to 360 channels, each gateway module supporting up to 180 channels.

• When using G.711 codec with 10 msec packetization time (without silence suppression) the gateway can support up to 240 channels, each gateway module supporting up to 120 channels.

• It is not recommended to use G.711 codec with 5 msec packetization time.

• When gateway is transmitting G.711 20 msec packets, it won't receive G.711 10 msec packets.

• Coder names in ini file are case sensitive.

• When changing gateway parameters from the Web Browser, the new parameters are permanently stored in flash memory only after the gateway is reset from the Web or after "Save Configuration" button is pressed.

• Not all parameters can be changed on-the-fly from Web Browser. Parameter that can be changed on-the-fly are noted with (*).. For changing other parameters, reset the board, using the Web browser reset button.



• When initiating a call from the PSTN side without a source phone number, the gateway inserts the configured channel number into the source phone number and sends it in an INVITE message to IP.

• Phone numbers from IP (SIP) to PSTN and vice versa are limited to 19 digits.

• Destination numbers on incoming calls from ISDN should not include asterisk (*) as the last digit of the number. If it does, the asterisk is deleted from the number, and the number without an asterisk is sent in INVITE SIP message to IP.

• If the Mediant 2000 gateway receives a BYE request simultaneously with sending a BYE request to the same call, it does not answer with “200 OK”.

• SNMP management doesn’t include SIP parameters. The ini file has to be used for Gateway configuration, using BootP & TFTP procedure or Web browser.

Mediant 2000 Gateway Functional Constraints


3.1 Mediant 2000 Gateway Clock Settings The gateway can be configured either to generate its timing signals from gateway internal clock source, or recover its clock from one of the E1/T1 trunks.

a. For using internal clock source the following parameters should be configured:

TDMBusClockSource = 1

ClockMaster = 1 (for all gateway trunks)

b. For using recovered clock the following parameters should be configured:

TDMBusClockSource = 4

ClockMaster = 0 (for all "slave" gateway trunks connected to Switch#1)

ClockMaster = 1 (for all "master" gateway trunks connected to Switch#2)

Assuming that the gateway will recover its internal clock from one of the "slave" trunks connected to Switch#1, and it will provide clock to Switch#2 on "master" trunks .

In addition it is required to define from which "slave" trunks ( the gateway will recover its clock):

PSTNAutoClockEnable = 1 (the gateway will automatically selec t one of the connected "slave" trunks) or

TDMBusLocalReference = # (Trunk index: 0 to 7, default = 0)



4. Mediant 2000 SIP Kit The Standard Kit Contents of the Mediant 2000/SIP Kit includes:

• User’s Manual.

• Image File for download such as ram.cmp.

• Sample Mediant_T1.ini and Mediant_E1.ini files.

• Default Call Progress Tone files (usa_tones.ini and usa_tones.dat).

• CAS protocol files for various signaling types, such as E_M_WinkTable.dat.

• The Excel™ utility for generation of the ini file.

• MIB library for SNMP browser.

• TDMUtil.exe utility for preparing Call progress tone.dat file.

• BootP/TFTP Configuration utility.

• These Release Notes.

Mediant 2000 Gateway Mediant 2000 SIP Kit


Reader’s Notes



5. Version History 5.1 Version 4.01

5.1.1 New Features

• Redirect Proxy server support (sending subsequent INVITE requests to the destination(s) indicated in SIP contact header(s).

• SIP Early media support, using 183 Session Progress (updating media sessions before the call is accepted). Both Generation and Receive.

• Registration to Proxy or to Registrar IP address.

• Modifying connection parameters in a call (RE-INVITE).

• Static NAT support (adding Global IP address in gateway configuration, and using it in SIP messages instead of local gateway IP address).

• Registration expired timer. Resume registration (Re-registration) if timeout is expired.

• Setting from ini file "IsUserPhone" parameter to control "user=phone" definition in SIP messages.

• Manipulation of destination numbers for both IPà PSTN and PSTNàIP calls.

• Setting of SIP signaling destination port (if other than 5600).

• Configuration of DTMF RFC 2833 and Fax bypass payload types from ini file.

• Setting Progress Indicator in ISDN/Q.931 messages .

• Optional play of ring-back tone to IP.

• Optional play of ring-back tone to PSTN.

• Interworking of ISDN/SIP and CAS/SIP release cause codes .

Mediant 2000 Gateway Version History

Previous Versions


5.1.2 Version 4.01 New Parameters Note: These parameters can be configured only from ini file, and not by using Web browser.


ini File Parameter Name Description

StaticNatIP Static NAT IP address. Global IPmedia server IP address. Should be defined if static Network Address Translation device (NAT) is used between the gateway and Internet.

EnableEarlyMedia 0 = Don't use Early Media 1 = Enable Early Media If enabled, the IPmedia server will send 183 Session Progress response (instead of 180 ringing), allowing media session to be established prior to the call being answered.

SIPDestinationPort SIP UDP destination port for sending SIP messages. Default = 5060

IsUserPhone 0 = don’t use "user=phone" string in SIP URL

1 = "user=phone" is part of SIP URL (default)

RegistrarIP IP address of Registrar server (optional). If not specified the gateway will register to Proxy Server

RegistrationTime Registration expired timeout (sec). The value is used in "Expires = " header. Typically value of 3600 is used, for registration for one hour. The gateway resumes registration before the timeout is expired.

PlayRBTone2Tel 0 = Don't play ringback tone to Telco 1 = Play ringback tone to Telco Default setting of this parameter depends on gateway type. By default, the CAS gateways play ringback tone to phone on receiving of 180/183 Ringing message. The ISDN gateways on the other hand won't play by default a ringback tone to Telco. Setting the "PlayRBTone2Tel" changes the default gateway behavior.

PlayRBTone2IP 0 = Don't play ringback tone to IP (default) 1 = Play ringback tone to IP When enabled, the gateway plays ringback tone to IP, immediately after sending 180/183 Ringing message.




ini File Parameter Name Description

NumberMapTel2IP a,b,c a = Phone number prefix b = Number of removed digits from number left side c = New prefix to be added, or "none" The parameter defines rule(s) for changing the destination phone number, received in Telcoà IP incoming call. Each rule is applied to a number that matches the "a" prefix. The "b" and "c" actions apply to a number that starts with "a" prefix. The actions apply to the number in the following order: "b" and than "c". The number manipulation rules, if defined, change the called phone number before routing the call (using "PREFIX = " paramete r). The new prefix ("c") can include any characters. Up to eight such rules can be defined in the ini file. The first (upper) rule is be applied first. The more general rules should be written as last rules. Examples: NumberMapTel2IP = 03,3,972 The rule deletes first 3 digits on numbers starting with "03", and replaces them with "972". 0312345 à 9722345. NumberMapTel2IP = 911,3,sos In this example 911 destination number is replaced by "sos" name.

NumberMapIP2Tel a, b, c a = Phone number prefix b = Number of removed digits from the number’s left side c = New prefix to be added, or "none" The parameter defines rule(s) for changing the destination phone number, received in IPàTelco outgoing call. Each rule is applied to the number that matches the "a" prefix. The prefix can include letters, such as "1abc". The "b" and "c" actions apply to a number that starts with "a" prefix. The actions apply to the number in the following order: "b" and then "c". Up to eight such rules can be defined in the ini file. The first (upper) rule is applied first. The more general rule should be written as the last rule. Examples: NumberMapTel2IP = 03,3,972 The rule deletes first 3 digits of numbers starting with "03", and replaces them with "972", 0312345 à 9722345


Previous Versions



ini File Parameter Name Description NumberMapTel2IP = 03,0,972111222333,0 All numbers starting with "03" are replaced with 972111222333.

FaxBypassPayloadType Fax Bypass RTP dynamic payload type Default =102

RTPDTMFRFC2833PayloadType The RFC 2833 DTMF relay dynamic payload type. Range: 96 to 99, 106 to 127; Default = 96 The 100 to 105 range is allocated for AudioCodes proprietary usage. Cisco™ are using payload type 101 for RFC 2833. The same payload type should be used for receive and for transmit.

5.1.3 Resolved Bugs

• VIA header line structure was inc reased to accept line of up to 150 characters.

• BYE request was not accepted OK in PRI gateway if Calling number was present in incoming PRI to IP call.

• Proper handling of Branch parameter in VIA header, assigning different Branch ID to new request (except CANCEL).

• SIP parser was modified to accept unknown SIP headers without hanging the gateway.

• In CANCEL message tag was removed from To header.

• Response to two or more Record-Route headers was fixed (missing ">" at end of generated record-route)

• Removed "Session-Expires" header. This keep-alive header caused ending of SIP session with third party proxy.



• SIP responses ( 1xx, 2xx, 4xx,...) are send now according to VIA header (according to RFC 3261 specifications)

• SDP domain name resolution to IP address, if domain name is used in "c = …" line

• SysLog configuration (Enable/Disable and IP address) can be changed on-the-fly from Web Browser.

• Filtering away RTP packets coming from wrong source IP address and port.

5.2 Version 4.0 GA

5.2.1 New Features

• Support for QSIG signaling (basic call).

• Support Domain names for Proxy and for Gateway

• DNS IP address resolution of domain names in VIA and Record Route headers. To use this feature DNS servers should be configured in the ini file or from the Web Browser.

• Sending ACK and BYE messages directly to Contact URL, if Record-Route was not present.

• Support of Loose and Strict routing

• Branch support in VIA header.

• Full support of 407 and 401 authentication responses

• Support of sub-headers: "User=phone" and "Transport=UDP"

• BYE request can be received instead of CANCEL

• The "fmtp" header was added in SDP (to support RFC 2833 DTMF relay)

• G.711 coder is supported with 5, 10 or 20 msec packetization time.


Previous Versions


• All string parameters (IP addresses, phone numbers, coder names) in ini file can appear without quotation marks (‘ ‘). File names still should appear with quotation marks.

• SysLog IP address can be changed on-the-fly from Web management

• Files can be downloaded to gateway directly from Web browser, using HTTP protocol.

5.2.2 Resolved Bugs

• Correct handling of Record-Route and Route headers (as per RFC 3261)

• Handling of large (above 1000 byte) SIP messages.

• Responding to SIP port, other than 5060

• IP TOS and IP Precedence fields can be configured from Web browser or from ini file.

• Silence compression can be configured from Web browser or from ini file.

• Automatic E1/T1 clock recovery (“TDMPSTNAutoClockEnable = 1”) is working now. If enabled, the gateway E1/T1 clock is recovered from one of the connected E1/T1 lines.

• When disconnecting a cable on E1/T1 span, the calls are disconnected only after a few seconds, enabling cable reconnection without closing the existing calls.

• Resolved Web browser hanging problem.

5.3 Version 4.0 Beta This is the first release of Mediant 2000/SIP digital gateway. For a detailed description of the product, refer to the AudioCodes Mediant 2000 Gateway SIP User’s Manual, Document #: LTRT-00688.



Reader's Notes


Previous Versions


US : AudioCodes Inc, 2890 Zanker Road, Suite # 200, San Jose, CA 95134.

Tel: 408-577-0488 - Fax: 408-577-0492

International: AudioCodes Ltd, 4 HaHoresh Street, PO Box 14, Yehud 56470, Israel.

Tel: +972-3-539 4000 - Fax: +972-3-539 4040

China: AudioCodes Ltd, Room 209 Tower A, Grand Pacific Building,

8A Guanghua Road, Chaoyang District, Beijing 100026, P.R.China

Tel: 86-10-6581-6622 - Fax: 86-10-6581-5599

Technical Support: [email protected]

Originalprinted on

recycled paperand available onCD or Web site

Documents

Mediant 2000™ SIP Gateway & TP-1610/SIP cPCI Board