02-19-2020 07:09 AM
Hi, I have the following environment where outgoing calls to external SIP telekom provider are failing.
Incoming calls from the ITSP are working perfectly.
CUCM => SIP trunk => CUBE => ITSP
The ITSP supports: G711A, G711U, G729a, G722.
My config is as follows:
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g722-64
voice class codec 2
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g711alaw
codec preference 4 g711ulaw
voice class codec 3
codec preference 1 g711ulaw
codec preference 2 g722-64
codec preference 3 g711alaw
codec preference 4 g729r8
voice class codec 4
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
dspfarm profile 1 transcode
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g722-64
codec g729br8
codec g729r8
maximum sessions 16
associate application SCCP
dial-peer voice 100 voip
description *Incoming from ITSP*
translation-profile incoming INCOMING_FROM_ITSP
session protocol sipv2
session transport udp
incoming called-number 3........$
incoming uri via ITSP
voice-class codec 1
voice-class sip profiles 1000
dtmf-relay sip-notify rtp-nte
dial-peer voice 200 voip
description *Outgoing to CUCMs*
destination-pattern 4...$
session protocol sipv2
session server-group 10000
voice-class codec 2
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0/0.7
voice-class sip bind media source-interface GigabitEthernet0/0/0.7
dtmf-relay rtp-nte
dial-peer voice 300 voip
description *Outbound to ITSP*
translation-profile outgoing OUTGOING_TO_ITSP
session protocol sipv2
session target sip-server
session transport udp
destination e164-pattern-map 2222
voice-class sip profiles 2000
dtmf-relay sip-notify rtp-nte
codec g711alaw
no vad
dial-peer voice 400 voip
description *Incoming from CUCM*
session protocol sipv2
incoming uri via CUCMs
voice-class sip bind control source-interface GigabitEthernet0/0/0.7
voice-class sip bind media source-interface GigabitEthernet0/0/0.7
dtmf-relay rtp-nte
no vad
And when I try an outgoing test call from one of the IP phones registered in the CUCM this is the debug I get (I´ve replaced the sensitive data with meaningful terms):
*Feb 19 12:45:16.109: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:Number2@CUBEs_Gi_IP:5060 SIP/2.0
Via: SIP/2.0/UDP CUCM_IP:5060;branch=z9hG4bK213b7562525ed
From: <sip:Number1@CUCM_IP>;tag=56087855~6a4d89ba-15e8-47e9-91bc-eed45860d755-63529998
To: <sip:Number2@CUBEs_Gi_IP>
Date: Wed, 19 Feb 2020 12:40:03 GMT
Call-ID: f1cf1900-e4d12ca3-1959c7-f904240a@CUCM_IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:CUCM_IP:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED
Cisco-Guid: 4056881408-0000065536-0000006392-4177798154
Session-Expires: 1800
P-Asserted-Identity: <sip:Number1@CUCM_IP>
Remote-Party-ID: <sip:Number1@CUCM_IP>;party=calling;screen=yes;privacy=off
Contact: <sip:Number1@CUCM_IP:5060>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 297
v=0
o=CiscoSystemsCCM-SIP 56087855 1 IN IP4 CUCM_IP
s=SIP Call
c=IN IP4 CUBEs_Gi_IP
b=TIAS:64000
b=AS:64
t=0 0
m=audio 8000 RTP/AVP 0 18 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Feb 19 12:45:16.113: //4/F1CF19000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP CUCM_IP:5060;branch=z9hG4bK213b7562525ed
From: <sip:Number1@CUCM_IP>;tag=56087855~6a4d89ba-15e8-47e9-91bc-eed45860d755-63529998
To: <sip:Number2@CUBEs_Gi_IP>;tag=42F18-195C
Date: Wed, 19 Feb 2020 12:45:16 GMT
Call-ID: f1cf1900-e4d12ca3-1959c7-f904240a@CUCM_IP
CSeq: 101 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-16.9.4
Session-ID: 1527fc57a8f45c5d9cf7f0fabf069885;remote=ac1dac794fdb5bb8a00b6e1236ade447
Content-Length: 0
*Feb 19 12:45:16.170: //4/F1CF19000000/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:Number2@CUBEs_Gi_IP:5060 SIP/2.0
Via: SIP/2.0/UDP CUCM_IP:5060;branch=z9hG4bK213b7562525ed
From: <sip:Number1@CUCM_IP>;tag=56087855~6a4d89ba-15e8-47e9-91bc-eed45860d755-63529998
To: <sip:Number2@CUBEs_Gi_IP>;tag=42F18-195C
Date: Wed, 19 Feb 2020 12:40:03 GMT
Call-ID: f1cf1900-e4d12ca3-1959c7-f904240a@CUCM_IP
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
As I understand the phone is offering those codecs to the CUBE:
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
But CUCM and CUBE are having a codec mismatch somewhere, the call is not even sent to the ITSP yet.
Additional data:
- my trunk in CUCM is configured for early-offer;
- my trunk in CUCM does not rely on an MTP; already tried that with same result;
- my trunk in CUCM does rely on a transcoder configured in CUBE;
- g711a is hardcoded on the outgoing dial-peer facing ITSP, I tried changing that with no luck;
- I tried putting all different four codec profiles I´ve got in the incoming dial-peer facing CUCMs with no luck. Also tried changing codec order, tried forcing everything to work with g711a, no luck.
- I checked phone region with trunk region relationship and its set to talk with g711 and g722.
Can someone see what I´m missing?
thank you,
02-19-2020 11:21 AM
02-20-2020 04:32 AM
02-20-2020 08:21 AM
Did you see the part where OP claimed to have tried voice class codec 2 already? If that's true, then I wonder why it didn't work for them.
"I tried putting all different four codec profiles I´ve got in the incoming dial-peer facing CUCMs with no luck."
02-21-2020 01:39 AM
I did see that but maybe not really took it in. I still think that should have worked, but would like to see the debug. Maybe it worked at the CUBE but got knocked back by the ITSP.
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide