Commit Graph

29 Commits

Author SHA1 Message Date
bossiel 3217795f85 - Allows setting max video down/up bandwidth
- Allows setting video motion rank
- Allows enabling/disabling STUN for ICE per SIP stack
2013-06-03 08:39:53 +00:00
bossiel 35e839e63d - Fix issue 231
- Add binaries for libfreetype and libfaac (win32)
- Update FFmpeg binaries fo TP system (win32)
2013-06-01 00:54:48 +00:00
bossiel 78094de99d Fix issue 268 2013-05-15 21:43:14 +00:00
bossiel c6ea8f7fab Fix issue 261 (Adds support for Opus audio codec)
Fix issue 262, issue 263 and issue 264
2013-05-07 04:55:21 +00:00
bossiel 181bc7b13d - Adds support for ZeroArtifacts (Perfect video quality)
- Better interop with WebRTC endpoints (better video quality)
- Lock-free on MediaSessionMgr for better performances on both audio and video
- Re-design the video jitter buffer for better CPU prefs and video quality. Request lost frames (RTC-NACK) as many times as required to deal with RTCP-losses. The FPS guesser is smarter.
- Fix issues on RTP timestamps on video pkts
- Update libsrtp binaries on Android and Windows (Use latest CVS)
- Better interop with other h264-rtp implementations (e.g. gstreamer, bria, cisco, polycom, lync...)
- Fix issue 233 (tinyNET does not compile on MAC + fix/patch)
- Fix issue 234 (tinyDAV does not compile on MAC)
- Fix issue 238 (iOS: Bad audio quality when audio/video call uses cpu intensive audio codec (e.g. g729 or speex))
- Fix issue 239 (Adds support for thread priority setting). Timers and audio/video threads now use high priority.
- Fix issue 242 (Hold/Resume fails when audio driver is opensl-es (Android))
- Fix issue 243 (PictureID in VP8 is not correct (only happens when there is overflow on the first 4 bytes))
- Fix issue 244 (Adds callbacks from codecs to session to signal IDR frames decoding)
- Fix issue 245 (Fail to decode h264 buffer)
- FIx issue 246 (Gnu Autotools: Detect support for monotonic timers in configure.ac)
2013-04-09 22:22:16 +00:00
bossiel 69c0e891df Allows relaying SRTP-Event (webrtc2sip) 2013-03-26 19:48:01 +00:00
bossiel 2e5d0d6038 Alert en-user when SIP TCP connection is lost 2013-02-27 01:58:06 +00:00
bossiel 5702098bc8 Adds support for Windows Phone 8 and Surface Pro 2013-02-17 18:56:03 +00:00
bossiel 017ac31766 Add support for DTLS-SRTP (rfc5764 and rfc5763) 2013-01-07 15:37:02 +00:00
bossiel c35b4e9255 Add support for Linux (webrtc2sip 2.0) 2012-12-03 03:11:21 +00:00
bossiel 348a1c7ee4 - Fix issue 59: Adds support for T.140 (rfc4103, rfc2198, rfc5194) - Thanks to IVèS (www.ives.fr/) for their contribution
- Make SWIG v2.0.8 the minimum version to generate bindings
2012-11-02 16:24:29 +00:00
bossiel 5428e86579 - Add supports for add-ons (plugins loaded from shared libs)
- Add WebRTC audio add-on for Linux systems
- Add OpenSL-ES audio add-on for Android 2.3+
- Fix RTC issue: BYE not sent when the call is ended
- Update Android build scripts to detect all toolchanins (up to 8b), include striping, make NDK r7c as the default toolchain
- Add support for Adaptive echo tail
- Update Speex libraries for Android
- Disable build for speakup-jb code when speex-jb is enabled
- Prevent SIGPIPE signal on iOS when using UDP
2012-09-03 06:41:33 +00:00
bossiel 31d1edf3ff - Allow setting RTC and RTCP-MUX options
- Fix issue on ICE negotiation when RTCP-MUX is disabled
- Make ICE negotiation smarter
2012-06-29 02:30:49 +00:00
bossiel d96205b245 - ICE (Interactive Connectivity Establishment): Full implementation of RFC 5245 for NAT Traversal
- 1080p (Full HD): all platforms supports full HD video negotiation. Off course it depends on your CPU and network bandwidth. The preferred video size could be changed from the QoS/QoS screen.
- Adaptive video jitter buffer: A video jitter buffer with advanced features like error correction, packet loss retransmission, delay recovery...
- RTP/AVPF profile as per RFC 4585
- RTCP: Full support for RTCP (3550) and many extensions such as: PLI (RFC 4585), SLI (RFC 4585), RPSI (RFC 4585), FIR (RFC 5104), NACK (4585), TMMBN (RFC 5104)...
- rtcp-mux as per 5761
- Negotiation of Generic Image Attributes in the SDP as per RFC 6236
- Source-Specific Media Attributes in SDP as per draft-lennox-mmusic-sdp-source-attributes-01
- Explicit Call Transfer as per 3GPP TS 24.629
2012-05-02 10:42:55 +00:00
bossiel c092aaf166 Add support for SRTP and some other cool stuff 2012-03-14 16:11:33 +00:00
bossiel 10722b880e Fix issue on session timers 2012-02-07 05:31:32 +00:00
bossiel bb40d86943 Fix bogue issue 95 2011-12-06 23:03:49 +00:00
bossiel 546fe46f0c Fix issue 75, issue 79 and issue 80 2011-10-26 00:32:57 +00:00
bossiel 78be530041 - Fix issue 56.
- simplify codec priority setting
2011-10-21 12:11:06 +00:00
bossiel 5bef56ad5e Fix issue 77 2011-10-13 22:38:16 +00:00
bossiel 28413584fe Begin integration with OpenTelePresence 2011-08-25 01:07:28 +00:00
bossiel f652615383 Allow setting codec priority\n Fix issue on RSeq=0 in 1xx reliable responses 2011-08-05 12:48:22 +00:00
bossiel 049f6025ff - Begin adding support for Google WebRTC
- This revision adds support for WebRTC's AEC and Noise Suppression
2011-08-01 05:14:45 +00:00
bossiel ab372e958c begin adding support for (real) arbitrary video size decoding (H.264) 2011-07-27 05:19:27 +00:00
bossiel 23ed807ada Allow the application to enabel/disable 100rel\nAllow forking/out-of-order NOTIFY requets 2011-06-23 17:13:16 +00:00
bossiel a62a9b2ada Add support for Noise suppression on iOS 2011-06-03 21:28:04 +00:00
bossiel 7f1240b44a Update v2.x 2011-05-12 22:14:22 +00:00
bossiel c91cf1bf1a Update v2.0 (begin adding support for RTCP) 2011-03-29 12:51:58 +00:00
bossiel 5448386f00 Add doubango v2.0 2011-03-25 09:38:07 +00:00