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| author | Alexander Traud <pabstraud@compuserve.com> | 2016-03-24 20:08:10 +0100 |
|---|---|---|
| committer | Alexander Traud <pabstraud@compuserve.com> | 2016-03-24 14:23:11 -0500 |
| commit | 81ce60f6d442e9e681bdfa72bc3d0204ad1cc744 (patch) | |
| tree | c52f3bf0a52b1cea0f2c187ddabb4bc2072780aa /include | |
| parent | d3af5320d43c73f0655ebaad1c818292c1250cc7 (diff) | |
chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers.
Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those
codecs, which the caller did not request/support. That fix was not complete
because on the second Session Timer all codecs were sent again. Some VoIP/SIP
clients interpreted that complete codec-list as a change in the SIP session.
Because of that, Asterisk did not send the RTP audio via NAT anymore which
created a non-audio scenario after the second Session Timer fired.
ASTERISK-24543 #close
Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66
Diffstat (limited to 'include')
0 files changed, 0 insertions, 0 deletions