quranteaching webrtc2sip issues

quranteaching / webrtc2sip

Automatically exported from code.google.com/p/webrtc2sip

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issues

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No audio from webrtc2sip to Chrome when attaching a legacy device

#66 GoogleCodeExporter opened 9 years ago
5
rtcweb breaker selecting wrong IP for SDP

#65 GoogleCodeExporter opened 9 years ago
6
Adds support foe TCP outbound

#64 GoogleCodeExporter closed 9 years ago
5
Adds support for "ping"/"pong"

#63 GoogleCodeExporter opened 9 years ago
1
Allow setting "stun_server" using the xml config file

#62 GoogleCodeExporter closed 9 years ago
1
How to configure this in Windows phone 8

#61 GoogleCodeExporter closed 9 years ago
1
DTLS role conflict with Firefox

#60 GoogleCodeExporter closed 9 years ago
3
ICE timeout when remote peer do not support rtcp-mux

#59 GoogleCodeExporter closed 9 years ago
3
Crash when Contact header is missing

#58 GoogleCodeExporter closed 9 years ago
3
Video line without formats

#57 GoogleCodeExporter closed 9 years ago
1
BYE never reach the caller

#56 GoogleCodeExporter opened 9 years ago
12
Windows phone 8 not accepting the dll of boghe .

#55 GoogleCodeExporter closed 9 years ago
1
Interop issue between webrtc2sip and JsSIP

#54 GoogleCodeExporter opened 9 years ago
0
Via header not stripped by webrtc2sip in 200ok response

#53 GoogleCodeExporter opened 9 years ago
2
No media

#52 GoogleCodeExporter closed 9 years ago
2
Contact header different between REGISTER and INVITE in rtcweb-breaker=yes case.

#51 GoogleCodeExporter opened 9 years ago
3
Bash-ism in autogen.sh

#50 GoogleCodeExporter opened 9 years ago
1
Chrome M24 and webrtc2sip : ICE issue ?

#49 GoogleCodeExporter opened 9 years ago
9
Media SSRC in RTCP-FIR from SIP-legacy to the browser is not correct

#48 GoogleCodeExporter closed 9 years ago
1
Add support for Firefox Nightly

#47 GoogleCodeExporter closed 9 years ago
2
webrct2sip + asterisk example wiki

#46 GoogleCodeExporter closed 9 years ago
2
chrome <-RTCWeb Breaker-> chrome do not work if media session not started on i200

#45 GoogleCodeExporter opened 9 years ago
0
Sound decode problem

#44 GoogleCodeExporter opened 9 years ago
3
FreeSWITCH crypto issue: "a=crypto in RTP/AVP, refer to RFC 3711"

#43 GoogleCodeExporter closed 9 years ago
1
Allow setting "rtp_symetric_enabled" using the xml config file

#42 GoogleCodeExporter closed 9 years ago
2
Allow setting preferred video size using xml config file

#41 GoogleCodeExporter closed 9 years ago
1
Media stops after a few seconds in a call

#40 GoogleCodeExporter closed 9 years ago
1
Please change file permissions of autogen.sh

#39 GoogleCodeExporter closed 9 years ago
2
Help needed starting webrtc2sip

#38 GoogleCodeExporter opened 9 years ago
3
Feature request: make webrtc2sip look for config.xml in default paths and/or add an option to set the path for the config.xml file

#37 GoogleCodeExporter closed 9 years ago
5
Must not use "a=mid:audio" without BUNDLE

#36 GoogleCodeExporter closed 9 years ago
1
Adds support for DTLS-SRTP

#35 GoogleCodeExporter closed 9 years ago
2
Copy headers defined in RFC 3840/41 (caller -> calle) when the RTCWeb Breaker is eanbled

#34 GoogleCodeExporter opened 9 years ago
0
Allows setting srtp mode in xml conf

#33 GoogleCodeExporter closed 9 years ago
1
rtp packets not forwarded

#32 GoogleCodeExporter opened 9 years ago
1
Adds 'samples' target in the Makefile

#31 GoogleCodeExporter opened 9 years ago
0
webrtc2sip fails to build: undefined references

#30 GoogleCodeExporter closed 9 years ago
7
netpacket/packet.h no such file on OS X lion

#29 GoogleCodeExporter closed 9 years ago
1
PTHREAD_MUTEX_RECURSIVE is undefined on debian

#28 GoogleCodeExporter closed 9 years ago
1
Crash on CentOS64 when browser (chrome) is refreshed without clean disconnect

#27 GoogleCodeExporter closed 9 years ago
2
Add support for opus audio codec

#26 GoogleCodeExporter closed 9 years ago
2
Add support for g722 audio codec

#25 GoogleCodeExporter closed 9 years ago
1
Allow logging to a file

#24 GoogleCodeExporter opened 9 years ago
0
Add support for Firefox native WebRTC implementation

#23 GoogleCodeExporter closed 9 years ago
3
Raise event from Doubango to webrtc2sip when SIP Transport stops

#22 GoogleCodeExporter opened 9 years ago
0
Raise event when network connection is down

#21 GoogleCodeExporter opened 9 years ago
0
Retrieve max FD_SIZE at runtime

#20 GoogleCodeExporter opened 9 years ago
0
Codec type mismatch when bypassing is enabled

#19 GoogleCodeExporter opened 9 years ago
0
Add support for audio encoding/decoding bypass

#18 GoogleCodeExporter opened 9 years ago
0
Add support for RTP timeout watcher

#17 GoogleCodeExporter opened 9 years ago
0