[Asterisk-code-review] chan sip: Fix lastrtprx always updated (asterisk[13])

[Matt Jordan]

Matt Jordan has posted comments on this change.

Change subject: chan_sip: Fix lastrtprx always updated
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Patch Set 2: Code-Review-1

(1 comment)

https://gerrit.asterisk.org/#/c/3032/2/channels/chan_sip.c
File channels/chan_sip.c:

Line 8601: 	if (fr && fr->frametype == AST_FRAME_VOICE) {
The RTP engine can return back more than just voice here when RTP is flowing. Any one of:
 * AST_FRAME_VOICE
 * AST_FRAME_VIDEO
 * AST_FRAME_TEXT
 * AST_FRAME_DTMF_BEGIN
 * AST_FRAME_DTMF_END

This patch would currently cause RTP timeouts if we were receiving video, text, or a whole mass of DTMF back to back.

If the RTP timeout is being reset due to RTCP, then filtering out responses that return ast_null_frame would be more effective and would not introduce regressions with non-audio streams.


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Gerrit-MessageType: comment
Gerrit-Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Evgeniy Tsybra <cjack at yandex.ru>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Evgeniy Tsybra <cjack at yandex.ru>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
Gerrit-HasComments: Yes