11 Commits (6fb4e0eaaa163465e5737c052d23add32a30ffd4)

Author	SHA1	Message	Date
Stefan Sayer	cd1752047b	fixes RTP_TIMEOUT bug in signaling only b2b situations ... - 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	19 years ago
Maxim Sobolev	58f33c8556	Second, much improved revision of adaptive jitter buffering patch. ... 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	19 years ago
Stefan Sayer	ce50f085ff	small cache optimisation. ssrc is not set! ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@170 8eb893ce-cfd4-0310-b710-fb5ebe64c474	19 years ago
Maxim Sobolev	0f59c71eb3	Implement real adaptive jitter buffering. Otherwise things like conferencing ... 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	19 years ago
Stefan Sayer	6d8f8d9645	enabled Id svn keyword property ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@145 8eb893ce-cfd4-0310-b710-fb5ebe64c474	19 years ago
Raphael Coeffic	9421bbe432	added dialout in conferencing module. ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@42 8eb893ce-cfd4-0310-b710-fb5ebe64c474	20 years ago
Raphael Coeffic	75a8aa318a	- added adaptive playout buffer with order statistic based delay estimation ... and adaptive WSOLA playout delay modification. - fixed serious bug in the RTP receiver. git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@41 8eb893ce-cfd4-0310-b710-fb5ebe64c474	20 years ago
Raphael Coeffic	c7a5180592	Many thanks to Rick van Rein for this patch. Here are his explainations to the path: ... The patch is developed on FreeBSD 4.11-RELEASE and gcc 2.95.4, but is expected to run on other BSD systems as well. Note that it is "not done" to replace the gcc in a BSD base system, hence the support for an older gcc. The changes are minor: * corrections in typing matters, including more header files * alternative handling option for varargs in #define for pre-GCC3 * lowering the default UNIX_PATH_MAX value to 104 instead of 108 * clearing memory intended to store socket addresses * a few remarks for FreeBSD-adopters and/or porters git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@39 8eb893ce-cfd4-0310-b710-fb5ebe64c474	20 years ago
Raphael Coeffic	555dd79d0a	reorganized session beginning: ... - sdp reply generation is now separated from the reply. - new onInvite. corrected a bug in getSession(callid,remote_tag). git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@22 8eb893ce-cfd4-0310-b710-fb5ebe64c474	20 years ago
Stefan Sayer	eabcee45c5	reenabled resynchronizing if TS out of range (in the case of clock skew or re-Invite) ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@19 8eb893ce-cfd4-0310-b710-fb5ebe64c474	20 years ago
Raphael Coeffic	37abd5379e	initial checkin new open source version. ... git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@5 8eb893ce-cfd4-0310-b710-fb5ebe64c474	20 years ago