[Asterisk-code-review] A non negotiated rtp frame causes call disconnection when there is a ... (asterisk[13])

[Joshua Colp]

Joshua Colp has posted comments on this change. ( https://gerrit.asterisk.org/c/asterisk/+/13846 )

Change subject: A non negotiated rtp frame causes call disconnection when there is a SSRC change
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Patch Set 4: Code-Review-1

(1 comment)

https://gerrit.asterisk.org/c/asterisk/+/13846/4/channels/chan_pjsip.c 
File channels/chan_pjsip.c:

https://gerrit.asterisk.org/c/asterisk/+/13846/4/channels/chan_pjsip.c@764 
PS4, Line 764: 		if (cur->subclass.format && ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), cur->subclass.format)
This will break the scenario where asymmetric RTP codec support is disabled and codec switching is permitted. I think the correct fix here is for this code to iterate through until it gets a voice frame. Only one voice frame will currently exist in a returned list, so document it. This frame is then used in the below media checks instead of "f".



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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec
Gerrit-Change-Number: 13846
Gerrit-PatchSet: 4
Gerrit-Owner: Paulo Vicentini <paulo.vicentini at gmail.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-Comment-Date: Mon, 02 Mar 2020 17:01:16 +0000
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