[asterisk-dev] [Code Review] Missing rtp changes in chan_skinny

[Dan Austin]

Russell wrote:
> On Wed, 2010-07-28 at 14:34 -0700, Dan Austin wrote:
>> I also see three other rtp using channels that
>> are missing the get_codec function-
>> 	chan_h323
>> 	chan_unistim
>> 	chan_multicast_rtp 
>> 
>> Should all three also be updated?

> That would be good.  It would also be good to just update
> main/rtp_engine.c to not blow up when that callback was not provided.

I am not sure I understand the code in the problem function:
ast_rtp_instance_early_bridge_make_compatible-

but perhaps this line (1389, and neighbor in current trunk):
if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec(c0)) {

should look like this:
if (audio_glue0_res == AST_RTP_GLUE_RESULT_REMOTE && (video_glue0_res == AST_RTP_GLUE_RESULT_FORBID || video_glue0_res == AST_RTP_GLUE_RESULT_REMOTE) && glue0->get_codec) {