[Asterisk-code-review] res pjsip sdp rtp: Properly set packetization on the codec (asterisk[13])

[Joshua Colp]

Joshua Colp has posted comments on this change. ( https://gerrit.asterisk.org/10495 )

Change subject: res_pjsip_sdp_rtp: Properly set packetization on the codec
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Patch Set 2: Code-Review-1

(2 comments)

https://gerrit.asterisk.org/#/c/10495/2/res/res_pjsip_sdp_rtp.c
File res/res_pjsip_sdp_rtp.c:

https://gerrit.asterisk.org/#/c/10495/2/res/res_pjsip_sdp_rtp.c@396
PS2, Line 396: 		if (codecs.framing) {
             : 			/* Copy the codecs framing ms even though there is no match */
             : 			ast_rtp_codecs_set_framing(ast_rtp_instance_get_codecs(session_media->rtp), codecs.framing);
             : 		}
Can you elaborate why exactly this is done? I'd expect the stream itself to be rejected, so I'm not sure why this would need to be done.


https://gerrit.asterisk.org/#/c/10495/2/res/res_rtp_asterisk.c
File res/res_rtp_asterisk.c:

https://gerrit.asterisk.org/#/c/10495/2/res/res_rtp_asterisk.c@5165
PS2, Line 5165: 	framing_inst1 = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance1));
              : 	framing_inst  = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance));
              : 	if (framing_inst1 != 0 && framing_inst != framing_inst1) {
              : 		return -1;
              : 	}
The bridge_native_rtp module already does checks on framing before allowing a bridge to occur - do they need to be changed there instead?



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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-MessageType: comment
Gerrit-Change-Id: Ic64b8fc6a140a93fbdb2f97550a40d0ff334e607
Gerrit-Change-Number: 10495
Gerrit-PatchSet: 2
Gerrit-Owner: Robert Cripps <rcripps at voxbone.com>
Gerrit-Reviewer: Jenkins2 (1000185)
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Comment-Date: Thu, 25 Oct 2018 13:45:58 +0000
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