Summary:ASTERISK-18614: Set Codec for MulticastRTP channel
Reporter:John Covert (jcovert)Labels:
Date Opened:2011-09-23 07:21:02Date Closed:2011-10-10 15:08:01
Versions: Frequency of
Environment:AllAttachments:( 0) chan_multicast_rtp.c.328209.patch
Description:As currently implemented in 1.8, a simple Dial(MulticastRTP/...)
command doesn't set the codec to something negotiated as the
"least common denominator" of all the listening phones, and I
wouldn't expect it to.

It will even leave the channel in signed linear in some cases.

The patch supplied allows you to set MULTICAST_RTP_CODEC
to ulaw (or whatever your phones support) before issuing the
DIAL command.
Comments:By: Leif Madsen (lmadsen) 2011-09-26 09:37:40.482-0500

Sorry, you're not going to be able to do this via a channel variable; that is a very old methodology that we avoid now. You'll probably need to add this to something like the CHANNEL() function.

By: Leif Madsen (lmadsen) 2011-09-26 09:37:49.791-0500

btw; thanks for the submission though :)

By: John Covert (jcovert) 2011-09-26 10:08:40.095-0500

I modelled this on the SIP_CODEC_OUTBOUND code.

You need to set this BEFORE doing the DIAL command, i.e. BEFORE the outbound channel even exists.

You need to be able to set it to something the inbound channel might not even support, and have transcoding take place in the outbound channel.

It could be done as part of the Dial String, i.e. MulticastRTP/address/already-used/Codec

Would that be acceptable?

The packet size also needs to be settable -- but one thing at a time.


By: Paul Belanger (pabelanger) 2011-10-10 15:07:34.568-0500

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines