[asterisk-dev] [Code Review] 2894: rtp_engine: ast_rtp_instance_early_bridge_make_compatible confusion over copy direction caused loss of rtp codec payloads
Matt Jordan
reviewboard at asterisk.org
Tue Oct 22 10:31:27 CDT 2013
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2894/#review9974
-----------------------------------------------------------
/branches/1.8/main/rtp_engine.c
<https://reviewboard.asterisk.org/r/2894/#comment19214>
Just to double check the logic change here:
Noting that ast_rtp_codecs_payloads_copy has the following signature:
void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance);
The original method calls copies the codecs from instance0 to instance1.
In your updated version, instance0 in the destination; instance1 is the source.
So the method cal now copies the codecs from the source (instance1) to the destination (instance0).
Is this what fixed the payload copy issue?
- Matt Jordan
On Sept. 30, 2013, 11:02 p.m., Scott Griepentrog wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2894/
> -----------------------------------------------------------
>
> (Updated Sept. 30, 2013, 11:02 p.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Bugs: ASTERISK-21464
> https://issues.asterisk.org/jira/browse/ASTERISK-21464
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> Testing for Issue 21464 (thanks Kevin!) turned up some odd behavior on codec negotiation, including loss of codec payloads. The arguments to ast_rtp_instance_early_bridge_make_compatible were not clearly indicating source and destination channel for the copy of codecs, which made it non-obvious that the arguments to ast_rtp_codecs_payloads_copy() were reversed. The result is that an offer containing 119 telephone event would be converted to 101 telephone event, for both directrtpsetup=yes and no.
>
>
> Diffs
> -----
>
> /branches/1.8/include/asterisk/rtp_engine.h 400206
> /branches/1.8/main/rtp_engine.c 400206
>
> Diff: https://reviewboard.asterisk.org/r/2894/diff/
>
>
> Testing
> -------
>
> Tested with 1.8 to prove that telephone-event payload code 119 is now being passed again (as it was in Asterisk versions prior to 1.8).
>
>
> Thanks,
>
> Scott Griepentrog
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20131022/604b18ff/attachment.html>
More information about the asterisk-dev
mailing list