- ... is too big.
- ... has a wrong header length.
Thanks to Andrei for that year-old-yet-not-applied patch!
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@221 8eb893ce-cfd4-0310-b710-fb5ebe64c474
- on RTP timeout, AmSession::onRtpTimeout is called
- session can be removed from media processor (detached)
by AmMediaProcessor::removeSession
- removing session from MediaProcessor and clearing audio
is AmMediaProcessor::clearSession (default action on
RTP error)
- added detach from media processor to conf_auth and ann_b2b
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@209 8eb893ce-cfd4-0310-b710-fb5ebe64c474
* new methods onSipRequest/onSipReply, the whole AmSipReply/AmSipRequest structure is passed to python
SipDialog
* state is changed only on reply to INVITE, not other requests
* made pin_collect work
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@189 8eb893ce-cfd4-0310-b710-fb5ebe64c474
Changes
-------
1. Make the AmJitterBuffer work with variable size RTP packets. Packet size
can be changed even during session (ex. Cisco in fax passthrough mode). Also
several improvements and fixes have been made to resyncronization logic.
2. Fix made to the AmPlayoutBuffer class to avoid reading chunks of size
larger than requested. This is required in cases when RTP packets contain
more or less data than internal frame size.
3. Small fix to AmRtpPacket class - replace the pointer to internal buffer
with offset in the buffer. This eliminates the nesessity to reparse the
packet each time the packet has been copied.
4. Replace the sample size field in amci_codec_t structure with two
functions - sampes2bytes and bytes2samples as that field did not allow to
specify sample size for LBR codecs (iLBC, gsm). This also brings ability
for codecs to determine the sample size at runtime (ex. iLBC).
5. Remove the sample size from amci_file_desc_t structure as it was used as
internal attribute of WAV files only and doesn't make much sense for other
file formats. Use the codec's ability to calculate sample size instead.
6. Parameter list for amci_inoutfmt_t.on_close() has been changed to give
ability to determine sample size in this file handler (WAV write_header
procedure requires this).
7. Fix gsm, ilbc, wav plugins and AmPlugin.c to reflect changes to amci.
Add corresponding samples2bytes and bytes2samples functions.
Caveats
--------
1. AmAdaptivePlayout class needs additional checking with RTP streams with
packets containing number of samples different from internal frame size
(for example 240 samples per packet in G711). Adaptive playout class
potentially may produce big packets of audio and therefore make the Conference
application work badly. The adaptive playout is used in Conference application
only and the application is working fine now with jitter buffer and without
adaptive playout. So I turned the adaptive playout off in the Conference app
as a workaround.
Developed by: Sippy Software, Inc.
Sponsored by: Digifonica Canada Limited
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@185 8eb893ce-cfd4-0310-b710-fb5ebe64c474
AmSIPEventHandler plugin, which receives replies to requests outside of a dialog.
* events can now be passed to session factories
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@158 8eb893ce-cfd4-0310-b710-fb5ebe64c474
module don't work very well on public networks with variable delay and jitter.
Still WiP, but should provide significant improvement over the current
behaviour.
Developed by: Sippy Software, Inc.
Sponsored by: Digifonica Canada Limited
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@156 8eb893ce-cfd4-0310-b710-fb5ebe64c474
* contact header is correctly sent in subsequent requests in dialout sessions
* contact header is not sent in BYE/CANCEL and positive replies to BYE
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@139 8eb893ce-cfd4-0310-b710-fb5ebe64c474
Raphael Coeffic
Stefan Sayer
Maxim Sobolev