Cisco 7911 with sip soft. Codec negotiation problem.

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Mar 4th, 2008
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Hello,


I have two Cisco 7911 phones registered to sip provider. I can make calls between endpoints and to pstn. But there is only one problem.


All calls are established using g.711.

In configuration file (SEP..cnf.xml) which is stored on TFPT server I use following command


<preferredCodec>g729a</preferredCodec>

Is that argument correct?


I have tried to change that to: g729, G.729A....and more.


Nothing has changed. I did test with Polycom with g711 codec disabled. Connection between 7911 i Polycom was established using g729. When all codecs were available on Polycom- connection used g711.


Logs shows following IP Phone behavior


(only fragments with codec)


CSeq: 101 INVITE


User-Agent: Cisco-CP7911G/8.3.0


Contact: <XXXXX;transport=udp>


Expires: 180


Accept: application/sdp


Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE


Supported: replaces,join,norefersub


Allow-Events: kpml,dialog


Content-Length: 322


Content-Type: application/sdp


Content-Disposition: session;handling=optional




v=0


o=Cisco-SIPUA 21303 0 IN IP4 10.23.204.4


s=SIP Call


t=0 0


m=audio 24176 RTP/AVP 0 8 18 116 101


c=IN IP4 10.23.204.4


a=rtpmap:0 PCMU/8000


a=rtpmap:8 PCMA/8000


a=rtpmap:18 G729/8000


a=fmtp:18 annexb=no


a=rtpmap:116 iLBC/8000


a=fmtp:116 mode=20


a=rtpmap:101 telephone-event/80

DBG 17:54:44.693792 JVM: 00


a=fmtp:101 0-15


a=sendrecv


-------



DBG 17:54:47.888491 JVM: sipSPICheckResponse: Response match: [email protected], cseq=102, cseq_method=INVITE

DBG 17:54:47.891294 JVM: Warning: No named events specified for fmtp attribute.

DBG 17:54:47.894618 JVM: Warning: No named events specified for fmtp attribute.

DBG 17:54:47.895469 JVM: Attribute label, level 1 instance 1 not found.

NOT 17:54:47.897748 DSP: CODEC[0] G.711 direction:2 cost:26 budget:87 available

NOT 17:54:47.898401 DSP: CODEC[1] G.729A or G.729AB direction:2 cost:41 budget:87 available

NOT 17:54:47.899025 DSP: CODEC[2] G.729 or G.729B direction:2 cost:41 budget:87 available

NOT 17:54:47.899568 DSP: CODEC[3] LINEAR 8 or 16kHz direction:2 cost:26 budget:87 available

NOT 17:54:47.900189 DSP: CODEC[4] G.722 direction:2 cost:32767 budget:87 NOT available

NOT 17:54:47.900720 DSP: CODEC[5] iLBC direction:2 cost:48 budget:87 available

NOT 17:54:47.901239 DSP: STREAM- GetCapableCodecList requestType:2 bitmap:0x2f

DBG 17:54:47.904874 JVM: Attribute ptime, level 1 instance 1 not found.

NOT 17:54:47.921501 DSP: STREAM- OpenIngressChan- ChanType 1, Remote (host 57cc8104, port 3eb0), medType 4, Period 20, VAD 0, TOS b8, stream (5, 5) --> chan 0


NOT 17:54:47.922213 DSP: STREAM- OpenIngressChan- mix (0, 0), dtmfpayloadtype 101


NOT 17:54:47.923992 DSP: Subtracted for CODEC[0] G.711 direction:1 cost:13 old budget:87

INF 17:54:47.928004 JVM: LibMT (vieoProcess.c) vieoSendOutgoingCASTMsg(), 8 bytes

DBG 17:54:47.951459 JVM: SIPSPIAddRouteHeaders: Route info not available; will not add Route header.

DBG 17:54:47.953241 JVM: sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, method=INVITE, to_tag=<SDei72499-1063472912-1204544073499>)

DBG 17:54:47.954359 JVM: sipTransportSendMessage: ccb <0>: config <87.X.X.X>:<5060> - remote <87.X.X.X>:<5060>

DBG 17:54:47.955796 JVM: sipTransportSendMessage: Sent SIP message: handle=<21>,length=<705>, message=

DBG 17:54:47.956378 JVM: ACK sip:[email protected]:5060;transport=udp SIP/2.0


Via: SIP/2.0/UDP 10.23.204.4:5060;branch=z9hG4bK346002ad



----


It seems like "preferred codec" is ignored. Cisco endpoint during negotiation starts witch g711 codec.


Do you know any solutions?

Thanks in advance

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pantoniuk Wed, 03/05/2008 - 04:22
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We have made some additional tests using Cisco IP Phone 7960. We have set preferred codec to g729 (we fill in configuration file exactly that argument).


Connection from 7960 endpoint to pstn was established using g729.


Connection from 7960 to 7911 was established using g711.


It seems that 7911 really doesnt understand argument in preferred codec line in config.


Do you have any reference to xml configuration file of new cisco phones (7XX1)?


Thanks,

Piotrek

Hi


I have seen this problem trying to get G.722 working between two Cisco phones registered to a 3rd party SIP Proxy.


The problem I believe is that in a CallManager environment, the CallManager rewrites the SDP offers based upon region configuration to ensure the correct codecs are used. So in non-CallManager environment, the SIP Proxy needs to replicate this behaviour. It seems like the SIP phones just ignore the preferred codec settings.

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