first version, wip.
todo:
- variable number of streams (now fixed to 3 - consumes many ports)
- symmetric RTP
- multi interfaces
- optimization: parse SDP only once, pass AmSdp object with request event
icmpwatcher has not been used for some time, as it is a
quite dangerous method of checking for the remote party
(in the beginning of a call due to delays in signalling,
the port on the remote end could for example still be not
open for some time)
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@1652 8eb893ce-cfd4-0310-b710-fb5ebe64c474
packets are treated the same as other RTP packets, they are all put into
a map with timestamp as key. However, some UA may use the same timestamp
for all rtpevent packets that represents a digit.
patch by Rui Jin Zheng rjzheng at boronetworks dot com
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@1090 8eb893ce-cfd4-0310-b710-fb5ebe64c474
* rtp packet parse called in RtpStream, so rtp keepalive packets are not disregarded in respect to rtp timeout
* rtp timeout made configurabe in sems.conf: dead_rtp_time
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@281 8eb893ce-cfd4-0310-b710-fb5ebe64c474
in the RTP stream. It is possible to disable this new behaviour using
single_codec_in_ok parameter in the sems.conf.
Developed by: Sippy Software, Inc.
Sponsored by: Digifonica Canada Limited
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@279 8eb893ce-cfd4-0310-b710-fb5ebe64c474
Adaptive jitter buffer can be compiled in instead by making with
WITH_ADAPTIVE_JITTERBUFFER=yes
(e.g. WITH_ADAPTIVE_JITTERBUFFER=yes make all)
Echo application switches with the star key the adaptive playout on and off.
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@225 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
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
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
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
- 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
Stefan Sayer
Raphael Coeffic
Maxim Sobolev