Reported by: wdecarne
Now that we pass through RTP timestamp information we need to make the allowed timestamp skew considerably less. There are situations where a source may change and due to the timestamp difference the receiver will experience an audio gap since we did not indicate by setting the marker bit that the source changed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
oscillating and incorrect data. Additionally, the RTT would sometimes report
negative values due to incorrect calculations.
(issue #9601, patch from davetroy)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Per comment from Dave Troy:
This adds back in some simple typecasting I had in an earlier version
which I realize now may be breaking things.
Issue #9554.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines
If an error occurs when reading from an RTP socket, and the error code does not
indicate that we should try again, then return NULL instead of a "null frame".
This will prevent Asterisk from trying over and over again, and eventually
causing the system to crash. (issue #8285, john)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
take the low 16 bits of the integer part, and the high 16 bits of the
fractional part. However, the code here was erroneously taking the low 16 bits
of the fractional part. It then shifted the result 16 bits down, so the result
was always zero. This fix makes it grab the appropriate high 16 bits, instead.
(issue #8991, pointed out by andre_abrantes)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
bridging can only be used when the DTMF modes don't match if the core is
monitoring DTMF in both directions. Then, the core will handle the translation.
Otherwise, this bridging method can not be used.
(issue #8936)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
something that video phones support in the RTP stream.
I now this is a big architectual change at this stage for 1.4, but decided it was needed
to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample
Issue 7679 in the bug tracker. Please test.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
