]> git.ipfire.org Git - thirdparty/asterisk.git/commit
chan_sip: Do not send all codecs on INVITE. 33/1533/1
authorAlexander Traud <pabstraud@compuserve.com>
Mon, 26 Oct 2015 16:42:03 +0000 (17:42 +0100)
committerAlexander Traud <pabstraud@compuserve.com>
Mon, 26 Oct 2015 16:46:48 +0000 (11:46 -0500)
commit1256aedf66b062c011959c422df1fa08e9f55522
treebff20a46b1c2f87774ee06e222fa602c3c10f9f9
parent31f13a1e930c62d49e713fca3f3978f26a9b2954
chan_sip: Do not send all codecs on INVITE.

Since version 13, Asterisk sent all allowed codecs as callee, even when the
caller did not request/support them. In case of dynamic RTP payloads, this led
to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the
intersection between the requested and the supported codecs is send again.

ASTERISK-24543 #close

Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287
channels/chan_sip.c