Preference out codec based on the region
Hi all
We have a CUCM cluster, connected to a CUBE connected to the ISP. All circuits are SIP:
CUCM - SIP SIP-CUBE - RTC.
The connection to the PSTN is centralized, but there are several offices. Through the region settings, all local traffic is set to G.711, but all WAN traffic is G.729, with the exception of faxes. The provider also supports G.711 and G.729 codecs. On incoming calls, apply the parameters of region, and most of the calls come as G.729, except those in the pool unit Faxes, which is connected to the region of faxes. The connection to the PSTN is also suitable for the region of the phone setting, so no transcoding occurs.
The outgoing however, we see that the connection to the PSTN is all that we put first in the class of codec, and if necessary, transcoding is done on the bridge. Is it possible negotiate the codec out according to the region of the phone settings?
We have now solved by adding a prefix to the number composed and corresponding to a different dial-peer for faxes, but I was wondering if it is possible otherwise.
Thanks in advance,
Tom
Tom after more research, I think I finally found how codecs are selected...
+++ Inbound call +++
Region between phone and cube is configured to use G711
inbound and outbound dial-peer configured to use the codec class voice with as prefered codec g729
+++ Cube receives invite to ITSPS all supported codecs and early offer +++
Received:
GUEST sip:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.50.49:5070; branch = z9hG4bKeao3hl307oi03u4716b0.1
From:; tag = 451687893-1338998863036-
To: 'codec test'[email protected] / * />
Call ID: [email protected] / * /
CSeq: INVITATION 558275167
Contact:
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, CONSULT, INFORM
Accept: multipart/mixed, application/media_control + xml, application/sdp
Support:
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 207
v = 0
o = BroadWorks 161384816 1 IN IP4 192.168.50.100
s = -.
c = IN IP4 192.168.50.100
t = 0 0
m = audio 15382 RTP / AVP-18 0 8 101
a rtpmap:101 telephone-event/8000 =
a = fmtp:101 0-15
a = ptime:20
a = fmtp:18 annex b = No.
+++ CUBE send invite to cucm with early supply and codecs on voice-class codec macthed +++
Envoy:
GUEST sip:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/TCP 172.16.5.10:5060; branch = z9hG4bK7954A198F
Remote-Party-ID:; part = call; screen = no; intimacy = off
From:; tag = 4C85AF62-DA1
TO:
Date: Wednesday, June 6, 2012 16:07:43 GMT
Call ID: [email protected] / * /
Supported: timer, resource-priority, replaces, sdp-anat
Min - SE: 1800
Cisco-Guid: 2499069035-2938638817-2515636045-1568562753
User-Agent: Cisco-SIPGateway/IOS - 12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: INVITE 101
Time stamp: 1338998863
Contact:
Expires: 180
Allow-events: kpml, telephone-event
Max-Forwards: 68
Content-Type: application/sdp
Content-Disposition: session; handling = required
Content-Length: 355
v = 0
o = CiscoSystemsSIP-GW-UserAgent 6494 4459 IN IP4 172.16.5.10
s = call SIP
c = IN IP4 172.16.5.10
t = 0 0
m = audio 16668 RTP / AVP-18 0 8 100 101
c = IN IP4 172.16.5.10
a G729/8000 rtpmap:18 =
a = fmtp:18 annex b = No.
a = rtpmap:0 PCMU/8000
a = rtpmap:8 PCMA/8000
a = rtpmap:100 X-NSE/8000
a = fmtp:100 192-194
a rtpmap:101 telephone-event/8000 =
a = fmtp:101 0-15
+++ After a couple to try and ring CUBE Gets an ok 200 of cucm with point of endpoints and IP codec +++
Received:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.16.5.10:5060; branch = z9hG4bK7954A198F
From:; tag = 4C85AF62-DA1
Up to:; tag=811681~ffa80926-5fac-4dd6-b405-2dbbc56ae9a2-477917860
Date: Wednesday, June 6, 2012 16:07:43 GMT
Call ID: [email protected] / * /
CSeq: INVITE 101
Allow: PROMPT, OPTIONS, INFO, BYE, ACK, CANCEL, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY
Allow-events: presence, kpml
Supported: replaces
Support: X-cisco-srtp-relief
Support: geolocation
Session-expires: 84600; refresher = UAS
Require: timer
Contact:
Content-Type: application/sdp
Content-Length: 214
v = 0
o = CiscoSystemsCCM-SIP 811681 1 IN IP4 10.100.10.50
s = call SIP
c = IN IP4 10.44.43.13
t = 0 0
m = audio RTP/AVP 0 101 27770
a = rtpmap:0 PCMU/8000
a = ptime:20
a rtpmap:101 telephone-event/8000 =
a = fmtp:101 0-15
+++ CUBE sends and ACK +++
Envoy:
SIP ACK:[email protected]/ * /: 5060; transport = tcp SIP/2.0
Via: SIP/2.0/TCP 172.16.5.10:5060; branch = z9hG4bK7954B2403
From:; tag = 4C85AF62-DA1
Up to:; tag=811681~ffa80926-5fac-4dd6-b405-2dbbc56ae9a2-477917860
Date: Wednesday, June 6, 2012 16:07:43 GMT
Call ID: [email protected] / * /
+++ CUBE then sends a 200 ok for ITSP with codec to use for the call endpoint.
Envoy:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.49:5070; branch = z9hG4bKeao3hl307oi03u4716b0.1
From:; tag = 451687893-1338998863036-
To: 'codec test'[email protected] / * />; tag = 4C85AF76-1
Date: Wednesday, June 6, 2012 16:07:43 GMT
Call ID: [email protected] / * /
CSeq: INVITATION 558275167
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-events: kpml, telephone-event
Remote-Party-ID:; part = called; screen = no; intimacy = off
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS - 12.x
Supported: timer
Content-Type: application/sdp
Content-Disposition: session; handling = required
Content-Length: 247
v = 0
o = 3682 3869 CiscoSystemsSIP-GW-UserAgent IN IP4 172.16.5.10
s = call SIP
c = IN IP4 172.16.5.10
t = 0 0
m = 25278 audio RTP/AVP 0 101
c = IN IP4 172.16.5.10
a = rtpmap:0 PCMU/8000
a rtpmap:101 telephone-event/8000 =
a = fmtp:101 0-15
a = ptime:20
+++ Received ACK then ITSP +++
002855: June 6 16:07:45.557: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP ACK:[email protected]/ * /: SIP-5060/2.0
Via: SIP/2.0/UDP 192.168.50.49:5070; branch = z9hG4bKhcdhmt20fg9gu7pf22b0.1
From:; tag = 451687893-1338998863036-
To: 'codec test'[email protected] / * />; tag = 4C85AF76-1
Call ID: [email protected] / * /
CSeq: ACK 558275167
Contact:
Max-Forwards: 69
Content-Length: 0
So in summary, here's how the codec is selected...
+++ INCOMING LEG +++
on the inbound leg...
The region setting will take effect no matter what you have configured on the voice-class codec to your incoming dial-peer. This is because when sending 200 ok CUCM will send the codec defined between the endpoint and the CUBE as the codec. CUBE then passes this ITSP.
+++ OUTGOING LEG +++
on the outbound direction...
Announced for ITSP codec will be used, ITSP being the decision that a codec use from the announced codec. So, here with the codec of voice of class, the preferred codec in the list is selected by the ITSP and this will be used for the call regardless of which is the region setting on the phone at the gateway of the cube.
If the codec is hardcoded... If its value g711, then g711 response will come from the ITSP and this codec will be used for the call.
If its g729, then it will be g729...
The outbound leg is so regardless of the setting of the region because it's the end that is to choose
which codec to use for the call.
If in both cases the codec selected by the end will determine which codec is used for the call
Inbound leg, end is CUCM... region takes effect
the outbound leg, end is ITSPS, prefered codec codec announced take effect... Simple!
So if you want to choose a codec for your outbound leg, you have either defined as favorite in your list of classes in voice or hard code it is the codec was announced as the prefferd to your ITSP
NB: for faxes, when entering, we use voice class codec and region settings (g711 is used for fax). As explained above, fax will use G711 because of its region at the gateway of cube settings
On the outbound leg, we did a similar config as yours. Prefix numbers for the called number, then configure dial-peer to use G711.
Please note the useful messages
"I am complete in God, God completes me.
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