[asterisk-dev] [Code Review] 2967: chan_sip: Fix RTCP port for SRFLX ICE candidates

[Matt Jordan]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2967/#review10056
-----------------------------------------------------------

Ship it!



branches/11/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2967/#comment19322>

    A space is needed between if and (


- Matt Jordan


On Oct. 28, 2013, 7:15 p.m., opticron wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2967/
> -----------------------------------------------------------
> 
> (Updated Oct. 28, 2013, 7:15 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-21383
>     https://issues.asterisk.org/jira/browse/ASTERISK-21383
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This corrects one-way audio between Asterisk and Chrome/jssip as a result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX ICE candidates.
> 
> 
> Diffs
> -----
> 
>   branches/11/channels/chan_sip.c 402109 
>   branches/11/include/asterisk/rtp_engine.h 402109 
>   branches/11/res/res_rtp_asterisk.c 402109 
> 
> Diff: https://reviewboard.asterisk.org/r/2967/diff/
> 
> 
> Testing
> -------
> 
> 
> Thanks,
> 
> opticron
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20131031/90440e18/attachment.html>