[Asterisk-code-review] res rtp asterisk / res pjsip: Add support for BUNDLE. (asterisk[master])

[Joshua Colp]

Joshua Colp has posted comments on this change. ( https://gerrit.asterisk.org/5981 )

Change subject: res_rtp_asterisk / res_pjsip: Add support for BUNDLE.
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Patch Set 3:

(2 comments)

https://gerrit.asterisk.org/#/c/5981/2/res/res_pjsip_sdp_rtp.c
File res/res_pjsip_sdp_rtp.c:

https://gerrit.asterisk.org/#/c/5981/2/res/res_pjsip_sdp_rtp.c@1396
PS2, Line 1396: 		media->conn = bundle_group_stream->conn;
              : 		media->desc.port = bundle_group_stream->desc.port;
> I don't quite understand, so Asterisk won't initially offer bundled streams
Each stream will be part of a bundle group but they won't be bundled yet. As a result the normal code above which has the RTP instance use its own transport information is used. Only if the other side accepts (or we accept) that the ->bundled field is set to 1 indicating it is actually bundled.


https://gerrit.asterisk.org/#/c/5981/2/res/res_pjsip_session.c
File res/res_pjsip_session.c:

https://gerrit.asterisk.org/#/c/5981/2/res/res_pjsip_session.c@449
PS2, Line 449: 				session_media->mid = ast_strdup(ast_codec_media_type2str(type));
> Yes reusing the id's for an answer or a reinvite makes sense, but for a new
I tried that but Firefox didn't raise the media streams. It only worked when I used the same mid as the others. I can revisit that though and see. It most certainly does work that way with Chrome.



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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: comment
Gerrit-Change-Id: I96c0920b9f9aca7382256484765a239017973c11
Gerrit-Change-Number: 5981
Gerrit-PatchSet: 3
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Benjamin Keith Ford <bford at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Comment-Date: Wed, 12 Jul 2017 16:47:53 +0000
Gerrit-HasComments: Yes
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