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author | Matthew Jordan <mjordan@digium.com> | 2013-03-08 03:54:38 +0000 |
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committer | Matthew Jordan <mjordan@digium.com> | 2013-03-08 03:54:38 +0000 |
commit | 12748bc7354d55c460e38f1001e9cb2692c529a6 (patch) | |
tree | d4777a075b35a566f9b771b86f763e79b26a0621 /main/asterisk.exports.in | |
parent | 3f0ea90ce65e93705347f47221862fc9e345a1b3 (diff) |
Don't reset the RTP address on a glare re-INVITE
Originally, way back in r201583, we added the alternate RTP address so
that the RTP engine would expect to receive audio from a new source
when a glare re-INVITE occurred. In r382589, we remove the alternate
RTP source, as the 'secret' probation mode allows for switching to a new
RTP source when a previous source stops sending RTP. At the time, it
seemed appropriate to set the RTP source based on the information in the
glared re-INVITE.
Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs
with no SDP - such as in a connected line update that glances - we'll set
the RTP source to an invalid address. In subsequent re-INVITE requests from
this Asterisk instance, we'll then send an invalid media address, which will
result in the remote side sending a 488. Whoops.
There isn't any need to reset the RTP source - if we're using strictrtp, we'll
simply synchronize to a new source when we stop getting packets from the old
one. If we aren't using strictrtp, then again there shouldn't be a problem.
Note that the Asterisk Test Suite's connectedline test caught this error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'main/asterisk.exports.in')
0 files changed, 0 insertions, 0 deletions