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New Member

Transcodig issue between SIP-TRUN and Autoattendant

Hi,

I have a doubt.

It's about transcoding.  I have a CME (router 2921) with AA.  When the CME receive an incoming call, i see that the call conect to the AA using

code G711ulaw.  I know this is the codec used for connecting AA.

But is not transcoding used to permit interaction between  g729 and g711.  The sip trunk is using g729abr8.

Which codec must be used to stablish the call?  g711ulaw or  g729abr8.

Because i want the call be stablish with  g729abr8.

Thanks..

8 REPLIES
New Member

Re: Transcodig issue between SIP-TRUN and Autoattendant

Hi,

It purely depends on your dial-peer configuration. The reason that your call is doing G.711ulaw because in the SIP invite, it might be advertising all codecs.

To force the call to be G.729, you need to ensure that you match a specific incoming dial-peer with "incoming called-number ." or specific called number. Ensure that the codec configure on this dial-peer is G.729. On the outbound leg, i.e. towards your AA you will have codec as G.711ulaw.

With this mismatch, it will then try to invoke the transcoder.

HTH.

Pratik

New Member

Re: Transcodig issue between SIP-TRUN and Autoattendant

Hi Patrick,

Thanks for your help.  I've already did what you told me.  Now i see that the call established between the CME and the SIP SERVER

is g729, and the side between CME and the AA is g711.

Here is the debug:

LAB_2900#sh call active voice compact
  A/O FAX T Codec       type        Peer Address       IP R:
Total call-legs: 4
       720 ANS     T8     g729r8      VOIP        P9392059953    10.234.146.76:20018   (sip server)
       721 ORG     T8     g711ulaw    VOIP        P5001   10.234.147.198:16908          (AA)
       722 ORG     T8     g729ar8     VOIP        P   10.234.147.193:2000                   (CME)
       724 ORG     T8     g711ulaw    VOIP        P   10.234.147.193:2000                  (CME).

LAB_2900#debug ccsip calls
SIP Call statistics tracing is enabled
LAB_2900# VCC, mtp1 index, mtp2 index, stream1, stream2 = 0 0 7 8 VCC, mtp1 index, mtp2 index, stream1, stream2 = 0 0 8 7
*Oct 25 12:02:30.519 GMT: //726/1AF05EF487ED/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x314AFD38
State of The Call        : STATE_ACTIVE
TCP Sockets Used         : NO
Calling Number           : 9392059953
Called Number            : 5001
Source IP Address (Sig  ): 10.234.147.193
Destn SIP Req Addr:Port  : 10.234.147.198:5060
Destn SIP Resp Addr
LAB_2900#:Port : 10.234.147.198:5060
Destination Name         : 10.234.147.198

*Oct 25 12:02:30.519 GMT: //726/1AF05EF487ED/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 8
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 10.234.147.193
Source IP Port    (Media): 18154
Destn  IP Address (Media): 10.234.147.19
LAB_2900#8
Destn  IP Port    (Media): 16906
Orig Destn IP Address:Port (Media): [ - ]:0

*Oct 25 12:02:30.591 GMT: //725/1AEF26A487E8/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x314B5460
State of The Call        : STATE_ACTIVE
TCP Sockets Used         : NO
Calling Number           : 9392059953
Called Number            : 9392055555
Source IP Address (Sig  ): 10.234.147.193
Destn SIP Req Addr:Port  : 10.234.146.76:5060
Destn SIP Resp Addr:Port : 10.234.146.76:506
LAB_2900#0
Destination Name         : 10.234.146.76

*Oct 25 12:02:30.591 GMT: //725/1AEF26A487E8/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g729r8
Negotiated Codec Bytes   : 20
Nego. Codec payload      : 18 (tx), 18 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 10.234.147.193
Source IP Port    (Media): 18000
Destn  IP Address (Media): 10.234.146.76
Destn  IP Port    (Med
LAB_2900#ia): 20024
Orig Destn IP Address:Port (Media): [ - ]:0

LAB_2900#
*Oct 25 12:02:41.507 GMT: //725/1AEF26A487E8/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x314B5460
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 9392059953
Called Number            : 9392055555
Source IP Address (Sig  ): 10.234.147.193
Destn SIP Req Addr:Port  : 10.234.146.76:5060
Destn SIP Resp Addr:Port : 10.234.146.76:5060
Destination Name         : 10.234.146.76

*Oct 25 12:02:41.507 GMT: //725/1AEF2
LAB_2900#6A487E8/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g729r8
Negotiated Codec Bytes   : 20
Nego. Codec payload      : 18 (tx), 18 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 10.234.147.193
Source IP Port    (Media): 18000
Destn  IP Address (Media): 10.234.146.76
Destn  IP Port    (Media): 20024
Orig Destn IP Address:Port (Media): [ - ]:0

*Oct 25 12:02:41.507 GMT:
LAB_2900# //725/1AEF26A487E8/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 200

*Oct 25 12:02:41.511 GMT: //726/1AF05EF487ED/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x314AFD38
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 9392059953
Called Number            : 5001
Source IP Address (Sig  ): 10.234.147.193
Destn SIP Req Addr:Port  : 10.234.147.198:5060
Destn SIP Resp Addr:Por
LAB_2900#t : 10.234.147.198:5060
Destination Name         : 10.234.147.198

*Oct 25 12:02:41.511 GMT: //726/1AF05EF487ED/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 8
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 10.234.147.193
Source IP Port    (Media): 18154
Destn  IP Address (Media): 10.234.147.198
D
LAB_2900#estn  IP Port    (Media): 16906
Orig Destn IP Address:Port (Media): [ - ]:0

*Oct 25 12:02:41.511 GMT: //726/1AF05EF487ED/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 200

Is it normal that stablish two session?

I am new in voice stuff.

Cisco Employee

Re: Transcodig issue between SIP-TRUN and Autoattendant

What codec is the AA advertising ? You can run 'debug ccsip message' and check the SDP in the SIP Messages for the codec advertised by each side.

Example :

10/25/2010 09:47:16.317 CCM|//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP
message from 10.25.0.19 on port 5060 index 302 with 1077 bytes:
SIP/2.0 200 OK
From:
<8130>;tag=330e0a89-2a55-452b-8bf8-4be44b71b16a-462459​62​
To:
<6387>;tag=91bfdf8-a190013-13c4-45026-340eb-48d7109e-3​40eb​
Call-ID: 5eec7900-cc518a63-11e-1600190a@10.25.0.22
CSeq: 101 INVITE
User-Agent: Cisco MCU 5.7.0.0.21 RVID534168404743475e5b0e58460e5015
Supported: replaces
Via: SIP/2.0/TCP 10.25.0.22;branch=z9hG4bKd965eb21e5e
Contact: <6387>
Content-Type: application/sdp
Content-Length: 597

v=0
o=RV-MCU 213228190 213228190 IN IP4 10.25.0.19
s=RV MCU Session
b=AS:384
t=0 0
m=audio 6084 RTP/AVP 0 9 18 101
c=IN IP4 10.25.0.19
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16
a=sendrecv
m=video 10012 RTP/AVP 104
c=IN IP4 10.25.0.20
b=AS:320
a=rtpmap:104 H264/90000
a=fmtp:104 profile-level-id=42E014
a=sendrecv
m=control 3337 tcp RvMcuNonStandard
c=IN IP4 10.25.0.19

New Member

Re: Transcodig issue between SIP-TRUN and Autoattendant

The AA is advertising this:

debug ccsip message output:

*Oct 25 14:58:59.578 GMT: //832/C27D18608917/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.234.146.76:5060;branch=z9hG4bKca22ki30dobhug03c1c0.1
From: "Polycom331 Polycom"<>9392059953@empresas.telefonica.pr;user=phone>;tag=SDekohe01-1117691625-1288033051686-
To: "User Pilot"<9392055555>;tag=F35255C-2090
Date: Mon, 25 Oct 2010 18:58:59 GMT
Call-ID: SDekohe01-68cc471429e3d48fb7274b2ac2a33c71-a0p02e3
Server: Cisco-SIPGateway/IOS-1
LAB_2900#2.x
CSeq: 845173269 PRACK
Content-Length: 0


*Oct 25 14:58:59.578 GMT: //832/C27D18608917/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.234.146.76:5060;branch=z9hG4bK9i1op520c8fhbj0op401.1
From: "Polycom331 Polycom"<>9392059953@empresas.telefonica.pr;user=phone>;tag=SDekohe01-1117691625-1288033051686-
To: "User Pilot"<9392055555>;tag=F35255C-2090
Date: Mon, 25 Oct 2010 18:58:59 GMT
LAB_2900#SDekohe01-68cc471429e3d48fb7274b2ac2a33c71-a0p02e3
CSeq: 845173268 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <9392055555>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 276

=0
o=CiscoSystemsSIP-GW-UserAgent 4229 9736 IN IP4 10.234.147.193
s=SIP Call
c=IN IP4 10.234
LAB_2900#.147.193
t=0 0
m=audio 19044 RTP/AVP 18 101
c=IN IP4 10.234.147.193
a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

The scenario is:

A SIP-SERVER (Broadsoft) connecte, via sip-trunk, to CME.  I call a pilot number 9392055555.

Here is the configuration for dial-peer:

dial-peer voice 939 voip
description Outbound to BS
translation-profile outgoing outgoing_clid
answer-address 9392055555
destination-pattern 939.......
session protocol sipv2
session target sip-server
voice-class codec 2 
dtmf-relay rtp-nte

Then i am using a ephone-dn  making call-forward all to 5001 (AA four digits extension).

Now when i use:

sho call active voice compac

This call legs are always stablished.  Is this normal?

  A/O FAX T Codec       type        Peer Address       IP R:
Total call-legs: 2
       814 ORG     T910   g729ar8     VOIP        P   10.234.147.193:2000
       816 ORG     T910   g711ulaw    VOIP        P   10.234.147.193:2000

Best regards,

New Member

Re: Transcodig issue between SIP-TRUN and Autoattendant

I think these are your transcoding legs. Is your transcoder registered on the same device?

Quick way to check is to execute following cmds when the call is active:

- sh call active voice brief

- sh sccp conn

- sh voip rtp conn

HTH

Pratik

New Member

Re: Transcodig issue between SIP-TRUN and Autoattendant

Hi,

Yes the transcoder is registered on the same device.

New Member

Re: Transcodig issue between SIP-TRUN and Autoattendant

Ok. Well in that case the output that you are noticing is correct.

You should notice 4 legs:

1) SIP Trunk

2) Xcoder - G.729 leg

3) Xcoder - G.711 leg

4) IP phone leg

R/Pratik

New Member

Re: Transcodig issue between SIP-TRUN and Autoattendant

Ok. Thanks..

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