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: callid=001ef728-519d0006-bda2a0ab-c2af000e@10.23.204.4, 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:X3@87.X.X.X: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
