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Defect #3017

open

re-invite problems?

Added by sean farrell about 9 years ago. Updated about 9 years ago.

Status:
New
Priority:
Normal
Assignee:
-
Category:
Sessions
Target version:
Start date:
09/25/2015
Due date:
% Done:

0%

Estimated time:
Keywords:

Description

Hello,

Why does the audio stop playing when I call and press 2? The purpose of pressing 2 is for a re-invite.

Full sip_trace.log attached for the session of the call and the sip2sip logs.

I see this in the sip2sip logs:
Packet 13 at 2015-09-25 20:34:00 from 209.12.167.61 to 81.23.228.150 (in)

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS 81.23.228.150:443;received=81.23.228.150;branch=z9hG4bKc587.fa05fde1.0
Via: SIP/2.0/UDP 76.74.151.123:5060;branch=z9hG4bK56059398-0000-4fa8ee65-be1113ee-7328055d
Record-Route: <sip:81.23.228.150:443;transport=tls;lr;r2=on;ftag=tlrx-62b21019-5605938c>
Record-Route: <sip:81.23.228.150;lr;r2=on;ftag=tlrx-62b21019-5605938c>
Call-ID: 3f9e502e12be48d186308cfbb470e79c
From: <sip:>;tag=tlrx-62b21019-5605938c
To: "sean" <sip:>;tag=a669407dffbc4c669dd089658553c298
CSeq: 10197929 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, gruu
Server: SIP2SIP 2.1.1 (Windows)
Content-Length: 0

Thanks,
Sean


Files

sip2sip.txt (16 KB) sip2sip.txt sean farrell, 09/25/2015 08:44 PM
sip_trace.txt (101 KB) sip_trace.txt sean farrell, 09/25/2015 08:44 PM
Actions #1

Updated by Saúl Ibarra Corretgé about 9 years ago

The codec line is changed from opus/48000/2 to juust opus/48000. This is detected by us as a codec change mid-call, which we don't support.

While we should support this, because it's essentially the same codec, I'm afraid it's complicated. You might have better luck asking ZipDX as well.

Actions #2

Updated by sean farrell about 9 years ago

Hello,

Thanks for the update. Disabling the opus codec all together will result in a
successful re-invite.

I had to specifically disable opus from the sip2sip app instead of rearranging
the priority, otherwise it continued to connect over opus. Is that expected?

Thanks,
sean

Actions #3

Updated by Saúl Ibarra Corretgé about 9 years ago

That's up to the recipient actually. Basically your SDP says "key, I support these codecs" and the recipient will say "cool, I choose that one".

Actions #4

Updated by sean farrell about 9 years ago

This issue may be closed. I've been able to call the VUC from sip2sip for two consecutive weeks, after disabling opus.

Special thanks to Michael Graves and Saúl Ibarra Corretgé.

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