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?
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
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
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:+firstname.lastname@example.org:5060 SIP/2.0
Via: SIP/2.0/UDP 184.108.40.206:5060;branch=z9hG4bK00B7ba16d2ba86f0cda
CSeq: 871999150 INVITE
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Content-Disposition: session; handling=required
o=Sonus_UAC 1696715527 1658374812 IN IP4 220.127.116.11
s=SIP Media Capabilities
c=IN IP4 18.104.22.168
m=audio 39502 RTP/AVP 8 18 0 9 101
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.
INVITE sip:+email@example.com SIP/2.0
Via: SIP/2.0/UDP 10.195.241.41:5060;branch=z9hG4bK43C5E5174F
Date: Sat, 11 Feb 2017 07:41:33 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
o=CiscoSystemsSIP-GW-UserAgent 4436 1370 IN IP4 10.195.241.41
c=IN IP4 10.195.241.41
m=audio 19588 RTP/AVP 18 0 9 101
c=IN IP4 10.195.241.41