cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
Announcements

Community Helping Community

409
Views
0
Helpful
11
Replies
Beginner

Problem video call from CME to CUCM

Hello! I have some problems with SIP trunk between CME and CUCM. When I cal from cme to cucm audio works fine, but video do not work. When I call from cucm to cme audio and video works fine. Both phones are sccp cisco 8941. On cme i configured rtp payload-type cisco-codec-video-h264  97. From "debug ccsip messages" I see that cme and cucm negotiate video ports and addresses. But from "show voip rtp conn" command i did not see video ports and address. Please help me!

11 REPLIES 11
Engager

Hi,

Hi,

Can you please share the output of debug ccsip messages of non-working video call? Also share show run.

Before changing the PT of h264, did you change the PT of fax-ack?

- Vivek

Beginner

Hi!

Hi!

Thank you for reply! I attached debug non-video call. 

Engager

Hi,

Hi,

SDP negotiation looks good.

Add following command under dial-peer 20000000;

dtmf-relay rtp-nte sip-notify sip-kmpl and see if it makes any difference.

Also share the output of debug ccsip messages of working video call.

- Vivek

Beginner

I added the command dtmf

I added the command dtmf-relay rtp-nte sip-notify sip-kmp, but whatever not video. I attached debug of successfully call.

Enthusiast

Can you share the voice class

Can you share the voice class codec 1 config, also if you can use codec transparent ?

For the working call, share the dial-peer which is being used.

Regards

Abhay Reyal

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle
Beginner

voice class codec 1

voice class codec 1

codec g729r8

codec g711alaw

codec g711ulaw

Incoming dial-peer:

dial-peer voice 444 voip
 description *** INCOMING PEER ***
 translation-profile incoming CUT_PREF_INCOMING
 rtp payload-type cisco-codec-fax-ack 112
 rtp payload-type cisco-codec-video-h264 97
 session protocol sipv2
 session transport udp
 incoming called-number 516[1-4]...
 voice-class codec 1 
 no vad

Beginner

Hi, Vivek!

Hi, Vivek!

Have you any ideas ?

Engager

Hi,

Hi,

SDP negotiation of non-working call seems good and similar to working call.

Can you please check with asymmetric payload full command? Simply remove rtp payload-type command and add 'voice-class sip asymmetric payload full' command in both incoming and outgoing dial peer. Using debug voice ccapi inout, please ensure that call is hitting the desired dial peer in both inbound and outbound direction.

- Vivek

Highlighted
Beginner

Hi, Vivek! No results.

Hi, Vivek!

No results.

Enthusiast

Share the debug ccsip

Share the debug ccsip messages with calling and called number. Also if you can make below change and see if that works or not.

no rtp payload-type cisco-codec-video-h264  97

rtp payload-type cisco-codec-fax-ack 111

rtp payload-type cisco-codec-video-h264 97

Make sure the binding is correct and if it is try to remove the binding and do it again. 

Regards

Abhay Reyal

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle
Beginner

Hi!

Hi!

Thank you for reply! I attached debug non-video call. 

CreatePlease to create content
Content for Community-Ad
FusionCharts will render here