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eburke authored Mar 19, 2015 and johan committed Mar 25, 2015

Signed-off-by: Johan Pascal <johan.pascal@belledonne-communications.com>

- setting in sip section srtp_crypto_suite in the configuration file
+ update ms

Ben Sartor authored Feb 05, 2015 and johan committed Mar 02, 2015

Signed-off-by: Johan Pascal <johan.pascal@belledonne-communications.com>

change the way payload type numbers are assigned, so that an application can support more payload type than the RTP profile table allows to contain.
Compliance with RFC3264 (offer answer model) is improved, by reusing numbers in case of reINVITEs.
Fix memory leaks
Move offer/answer related tests into a new test suite.

tester automatically creates unique accounts on flexisip server before running tests. This allows several developer to run the test suite simultaneously !

Normally this should not trigger any notification.
Fix bug allowing two incoming calls to be notified if ICE is used.

add linphone_call_media_in_progress() method for app to easily check that ice has finished or not its processing.
Update GTK app accordingly, so that adding video is no longer possible while ICE is in progress.

rate!=8000 and rate!=16000
no hardware AEC
AEC required (thus software)
webRTC AEC is used
not opus (because opus can accept 16khz in input)

Inactive streams are now allowed between active streams in the SDP.

add new tests.

- vbr codecs can automatically have different output bitrates depending on whether video is used and/or allowed total output bandwidth
- application can specify an output IP bitrate for a given codec, which allows to control the quality of vbr codecs.
Note: a belle-sip upgrade is required to fix a bug around channels parsing in rtpmap.

add ugly hack to allow older versions of linphone to call new versions with opus.

Implies a lot of refactoring in streams management.