[Asterisk-code-review] Add rtcp-mux support (asterisk[13])

[Richard Mudgett]

Richard Mudgett has posted comments on this change. ( https://gerrit.asterisk.org/5138 )

Change subject: Add rtcp-mux support
......................................................................


Patch Set 3: Code-Review-1

(4 comments)

https://gerrit.asterisk.org/#/c/5138/3/include/asterisk/res_pjsip.h
File include/asterisk/res_pjsip.h:

PS3, Line 675: 	/*! Use RTCP-MUX */
             : 	unsigned int rtcp_mux;
This location for the rtcp_mux option is fine only for master.  Were you going to crate a version for v13 and v14 that doesn't  break ABI?


PS3, Line 759: 	unsigned int asymmetric_rtp_codec;
             : };
The rtcp_mux option needs to go here to not break ABI on v13 and v14.


https://gerrit.asterisk.org/#/c/5138/3/res/res_pjsip.c
File res/res_pjsip.c:

PS3, Line 952: 						on the same port. This leaves the demultiplexing logic in the application rather than
             : 						at the transport layer. This option is useful when interoperating with WebRTC endpoints
"This shifts the demultiplexing logic to the application layer rather than the transport layer."


https://gerrit.asterisk.org/#/c/5138/3/res/res_rtp_asterisk.c
File res/res_rtp_asterisk.c:

PS3, Line 4562: 	m = packet[1] & 0x80;
              : 	pt = packet[1] & 0x7F;
              : 	if (m && pt >= 64 && pt <= 95) {
This can be simplified to:
if ((0x80 | 64) <= packet[1] && packet[1] <= (0x80 | 95))


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Gerrit-MessageType: comment
Gerrit-Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: 13
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-Reviewer: Matthew Fredrickson <creslin at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Sean Bright <sean.bright at gmail.com>
Gerrit-HasComments: Yes