Re-Invite / Upgrading Media

Asterisk Asterisk SIP
1

Does Asterisk support upgrading the media when a Re-Invite comes in? The flow is as follows:

  • INVITE comes in from a trunk with Audio only in the SDP
  • After the call is ANSWERED, the remote side sends a Re-INVITE with Audio & Video in the SDP.

Looking at the Wireshark, it seems like Asterisk responded to the RE-INVITE.

Following is the SDP that was in the RE-INVITE

v=0
o=inbcf 1936544892 2 IN IP4 172.25.52.69
s=inbcf_1971237465
c=IN IP4 172.25.52.69
t=0 0
m=audio 20076 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=video 20078 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801f

Following is the OK which Asterisk responded with

v=0
o=root 342471461 342471462 IN IP4 172.25.54.153
s=Asterisk PBX 17.8.1
c=IN IP4 172.25.54.153
t=0 0
m=audio 15504 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv
m=video 0 RTP/AVP 96

Looks like there was a similar issue reported which seem to have been resolve. However, I’m experiencing the same behavior.

Thanks.

2

Which version? It certainly used to require all streams to be present in the initial INVITE, but it is just possible that has changed.

Which channel driver?

3

Using chan_sip (I know I should be upgrading to pjsip, hopefully will get there soon)
Asterisk version 17.8.1

4

I think you can safely say that chan_sip cannot add streams mid call.

5

So if I were to use pjsip, would it be able to handle adding streams mid call?

6

PJSIP allows adding/removing streams mid-call. However, you are using a security fix only version of Asterisk that may not have received fixes/changes in that area, and certainly will not in the future.

7

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