and some patches (all disclaimed).
- Don't change RTP properties if we reject a re-INVITE
- Don't add video to an outbound channel if there's no video on the inbound channel
- Don't include video in the "preferred codec" list for codec selection
- Clean up and document code that parses and adds SDP attachments
Since we do not transcode video, we can't handle video the same way as audio. This is a
bug fix patch. In future releases, we need to work on a solution for video negotiation,
not codecs but formats and framerates instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
update iax2_indicate to pass control frame payload to the connected channel
add an API call for sending an indication with payload, and use it for control frames with payload
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
on the frame counters. Document it in the header file.
- provide a single exit point for a function;
- mark XXX some unclear parts of the code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
wrappers around the basic 'say' functions, and redeclare these
wrappers as ordinary functions rather than function pointers.
This way, alternative implementations of the 'say' functions
will only have to implement the basic functions and not the
wrappers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- use ast_channel_lock/unlock in a few places
- comment some dubious pieces of code
- use memset to zero a buffer
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
don't transcode via SLINEAR when the option is enabled but there is a direct path from the source to the destination
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- handle immediately failures in ast_request();
This removes the need for checking 'chan' multiple times afterwards.
- handle immediately failures in ast_call(), by moving the one-line
case at the top of the 'if' statement;
- use ast_strlen_zero in several places instead of expanding it inline;
- make outstate always a valid pointer;
On passing mark an unclear statement and replace a magic number
with sizeof(tmp).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- bring the short case at the top of an 'if' statement
(also fix misformatting)
- replace several 'if' with the '?' operator;
- invert the condition on an 'if' to reduce the nesting depth
(reindentation to be done later).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
of their '_full()' version, so we can remove the replicated
implementation and, especially, the risk that they get out of sync.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Update lock.h with definitions of ast_channel_lock, ast_channel_unlock and ast_channel_trylock
- Convert some functions (but not all) in channel.c
- Fix some bugs in chan_sip.c
- Convert rest of chan_sip.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when you have channel locking issues.
(Part of the SIP transfer patch, where I had a *lot* of
channel locking problems)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
