CUBE with Webex Calling SDP

[Eike Scholz-Janotte]

Hi 

I have a vCube running Version 17 and all is configured. Sip Trunk to Webex Calling and another to our PSTN provider (also SIP trunk).

Now we were testing some calls and have a strange behavior.

The received Invite from Webex contains three lines for the video codec in sdp

Received
a=rtpmap:112 H264/90000
a=fmtp:112 profile-level-id=640c16;packetization-mode=1;max-fs=3600;max-mbps=108000
a=rtpmap:111 H264/90000
a=fmtp:111 profile-level-id=428016;packetization-mode=1;max-fs=3600;max-mbps=108000
a=rtpmap:110 H264/90000
a=fmtp:110 profile-level-id=428016;packetization-mode=0;max-fs=3600;max-mbps=108000

The Invite sent to the provider contains also three lines but two lines have the same value, which will result in a 400 bad request

Sent
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=640C16;packetization-mode=1;max-mbps=108000;max-fs=3600
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=428016;packetization-mode=1;max-mbps=108000;max-fs=3600
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428016;packetization-mode=0;max-mbps=108000;max-fs=3600

It seems that the cube assume "..it is actually the same stuff so just use the same rtmap again.."

I was trying to configure a video codec on the codec list, but this didn't change anything. Modifying the SDP a= header is not possible as I understood from cisco documentations.

Did anyone had the same issue and resolved it or have an idea how to fix it?

Thanks

Eike  

[Nuno Melo · Accepted solution]

Assuming your provider does not support video codecs witch is usual on PSTN , you can use the following to completely strip video codecs:

Device(config)# voice service voip
Device (config-voi-serv)# sip
Device (config-voi-sip)# audio forced
Device (config-voi-sip)# end

This will make the cube strip any video information on invites towards isp, if you don't want to disable video cube wide and can make it dial-peer specific using voice-class sip audio forced on the dial-peer

[Carlo Poggiarelli]

Hi Eike,

on dial-peer pointing your provider , add this.

no voice-class sip pass-thru content custom-sdp

It strips custom sdp message so your provider can receive a standard formatted sdp message.

HTH

Cheers

Carlo 

[Eike Scholz-Janotte]

Thank you guys

I did both of the commands already as workaround. The command "no pass-thru content custom-sdp" was already applied at the tenant section. Audio forced solved it, I was just curious how this can happen.

Eike