-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4093/
-----------------------------------------------------------
(Updated Dec. 1, 2014, 12:51 p.m.)
Status
------
This change has been marked as submitted.
Review request for Asterisk Developers.
Changes
-------
Committed in revision 428708
Bugs: ASTERISK-24274
https://issues.asterisk.org/jira/browse/ASTERISK-24274
Repository: Asterisk
Description
-------
Currently SLIN12, SLIN24, SLIN32, SLIN44, SLIN48, SLIN96 and SLIN192 are found
not working with SIP. The following error will be thrown if one of those codecs
is used: chan_sip.c:10718 process_sdp: No compatible codecs, not accepting this
offer!
What I think the issue is that the codec format isn't being included in the SDP
media attributes when one of those codecs is used. Please refer to
ASTERISK-24274 for more details. This change updates the main/rtp_engine.c and
main/frame.c to ensure all these codecs are supported.
Note: SLIN and SLIN16 are working fine.
Diffs
-----
/tags/12.4.0/main/rtp_engine.c 425756
/tags/12.4.0/main/frame.c 425756
Diff: https://reviewboard.asterisk.org/r/4093/diff/
Testing
-------
Specified SLIN48 codec in sip.conf of two Asterisk servers. Used AMI to
originate a call from Server A to Server B and then put Server B into a
conference hosted in Server A. The above mentioned error was no longer reported
and the conference was working as expected.
Thanks,
Frankie Chin
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
