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asterisk/channels
Alexander Traud 81ce60f6d4 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
2016-03-24 14:23:11 -05:00
..
2012-06-15 16:20:16 +00:00
2012-06-15 16:20:16 +00:00