[Asterisk-code-review] SDP: Ensure SDPs "merge" properly. (asterisk[master])

[Mark Michelson]

Mark Michelson has posted comments on this change. ( https://gerrit.asterisk.org/5415 )

Change subject: SDP: Ensure SDPs "merge" properly.
......................................................................


Patch Set 2:

(2 comments)

https://gerrit.asterisk.org/#/c/5415/2/main/sdp_state.c
File main/sdp_state.c:

Line 578: 			joint_stream = ast_stream_alloc("dummy", new_stream_type);
> will the dummy stream end up in the channel's streams in the end?
Yes, most likely. And that's probably not a good thing.

The idea here is that you want the stream to be present so that SDPs generated from the capabilities include an m-line with a 0-port*. However, exposing that to users of the API is likely not a good thing. Got a good idea for how to work with this?


*Come to think about it, that's not actually being done here yet, so that's something that will need to go into the next revision as well.


Line 806: 		if (state_stream->instance) {
> This should only occur if audio or video
I counter that only an audio or video stream will have an RTP instance due to the other changes you have recommended.


-- 
To view, visit https://gerrit.asterisk.org/5415
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: comment
Gerrit-Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-HasComments: Yes