[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 5: Code-Review-1

(2 comments)

https://gerrit.asterisk.org/c/asterisk/+/13846/5/channels/chan_pjsip.c 
File channels/chan_pjsip.c:

https://gerrit.asterisk.org/c/asterisk/+/13846/5/channels/chan_pjsip.c@768 
PS5, Line 768: 
You can do if (!cur) { return f; } here - and remove the checks for cur below. This matches existing behavior and still does the voice frame check.


https://gerrit.asterisk.org/c/asterisk/+/13846/5/channels/chan_pjsip.c@811 
PS5, Line 811: 	//legacy
             : 	if (f->frametype != AST_FRAME_VOICE) {
             : 		return f;
             : 	}
This can be removed



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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec
Gerrit-Change-Number: 13846
Gerrit-PatchSet: 5
Gerrit-Owner: Paulo Vicentini <paulo.vicentini at gmail.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-Comment-Date: Tue, 03 Mar 2020 10:42:58 +0000
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