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Participant

Voice Class Codec and Regions

Looking to confirm that when voice classes are used on the router the codec negotiated will be decided in the regional relationship regardless of the priority in the voice class.

Can anyone confirm this?

Thanks

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5 REPLIES 5
VIP Advisor

Re: Voice Class Codec and Regions

The codec negotiation will be based on the priority listed in the class.
The 1st match is used
Participant

Re: Voice Class Codec and Regions

So if the carrier is sending an early offer g711 and my voiceclass has P1-729 and P2-711, the region is set to 64k between Cube and phone you are saying the call will be 729 because that is the first priority in the voice class?

I have heard conflicting reports that this call would actually be negotiated at 711

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Cisco Employee

Re: Voice Class Codec and Regions

I believe 729 will be used only if the CUBE invokes a LTI, else it has to be 711 since there would be a codec mismatch resulting in a call failure.

Nipun Singh Raghav
"We cannot solve our problems with the same thinking we used when we created them"
VIP Advisor

Re: Voice Class Codec and Regions

It will use G711 because your provider is sending G711 in the SDP. CUBE and
CUCM will alway avoid using xcoder as possible. The initiator is
controlling the codec to be used and responder will try to match it as
possible. Otherwise xcoder will used.
VIP Mentor

Re: Voice Class Codec and Regions

The codec negotiated will be a combination of the codec listed in your voice class and the bit rate set on the region. The offer present in the call or the direction of the call will also impact on the codec that will be used. CUBE always uses the concept of codec filtering before sending out a request

Ex:

voice class codec: G711u-pr1, G711alaw-pr2

Assuming ITSP sends only G711alaw and the region allows 64k, G711alaw will be negotiated for this call.

 

However supposing you are making an outbound call and you offer both G711ulaw-pr1 and G711a-pr2, and your ITSP can support both, then G711ulaw should be used. Your preferred codec should be honoured.

In the example you gave G711 will be used because thats the only codec supported by the UAC.

 

CUBE will always filter codecs out based on what is configured on the dial-peer. Another example below

 

1. outboud dial-peer to CUCM is configured with

voice class codec 1

codec pref 1 g722-64

codec pref 2 g711u

codec pref 3 g729r8

 

2. Inbound INVITE from ITSP is shown belowReceived:
INVITE sip:+37205555585@120.66.80.118:5060 SIP/2.0
Via: SIP/2.0/UDP 34.205.187.90:5060;branch=z9hG4bK00B7ba16d2ba86f0cda
From: <sip:+449966627386@34.205.187.90>;tag=gK002a38eb
To: <sip:+37205555585@120.66.80.118>
Call-ID: 541080909_16750612@34.205.187.90
CSeq: 871999150 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay,  multipart/mixed
Contact: <sip:+449966627386@34.205.187.90:5060>
P-Preferred-Identity: <sip:+449966627386@34.205.187.90:5060>
Supported: timer,replaces
Session-Expires: 1800
Min-SE: 90
Content-Length:   343
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 1696715527 1658374812 IN IP4 34.205.187.90
s=SIP Media Capabilities
c=IN IP4 34.205.187.89
t=0 0
m=audio 39502 RTP/AVP 8 18 0 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20

3. Outbound INVITE to CUCM ( observe that CUBE has filtered out any codec not present in the voice class attached to the dial-peer) but it will maitain the order of preference once filtering is done.

Sent:
INVITE sip:+37205555585@10.195.126.201 SIP/2.0
Via: SIP/2.0/UDP 10.195.241.41:5060;branch=z9hG4bK43C5E5174F
Remote-Party-ID: <sip:+449966627386@10.195.241.41>;party=calling;screen=no;privacy=off
From: <sip:+449966627386@10.195.241.41>;tag=D8E38B73-223F
To: <sip:+37205555585@10.195.126.201>
Date: Sat, 11 Feb 2017 07:41:33 GMT
Call-ID: 57F967E4-EF6411E6-B2FCF530-ECD03D62@10.195.241.41
Supported: timer,resource-priority,replaces,sdp-anat
Require: 100rel
Min-SE:  1800
Cisco-Guid: 1475932947-4016312806-3002529072-3973070178
User-Agent: Cisco-SIPGateway/IOS-15.5.3.S3
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1486798893
Contact: <sip:+449966627386@10.195.241.41:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 342

v=0
o=CiscoSystemsSIP-GW-UserAgent 4436 1370 IN IP4 10.195.241.41
s=SIP Call
c=IN IP4 10.195.241.41
t=0 0
m=audio 19588 RTP/AVP 18 0 9 101
c=IN IP4 10.195.241.41
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=fmtp:9 bitrate=64
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

 

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