hjc123186 webrtc2sip issues

hjc123186 / webrtc2sip

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

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issues

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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
Detect max internal socket buffer at runtime

#16 GoogleCodeExporter opened 9 years ago
0
Adds support for RTCP-NACK forwarding when video jb is disabled

#15 GoogleCodeExporter opened 9 years ago
0
Use non-zero mask key for WebSocket messages

#14 GoogleCodeExporter opened 9 years ago
0
Disconnect old sockets

#13 GoogleCodeExporter closed 9 years ago
1
tinyWRAP_wrap.cxx:308: error: invalid conversion from 'void**' to 'JNIEnv**'

#12 GoogleCodeExporter closed 9 years ago
2
Bad Request - Not following indicated Service-Routes

#11 GoogleCodeExporter opened 9 years ago
1
SIP/2.0 401 Unauthorized - Challenging the UE

#10 GoogleCodeExporter closed 9 years ago
2
error Webrtc2sip with IMSDroid /v2.0.509

#9 GoogleCodeExporter closed 9 years ago
1
running webrtc2sip

#8 GoogleCodeExporter closed 9 years ago
1
Adds support for draft-hixie- thewebsocketprotocol-76

#7 GoogleCodeExporter opened 9 years ago
0
Add support for Bowser (Ericsson's WebRTC implementation)

#6 GoogleCodeExporter opened 9 years ago
0
Content-Lenght is NOT required in SIP over WebSocket

#5 GoogleCodeExporter closed 9 years ago
1
Connect via TLS transport

#4 GoogleCodeExporter closed 9 years ago
1
Connect TLS

#3 GoogleCodeExporter closed 9 years ago
2
Fail to build tinySIP

#2 GoogleCodeExporter closed 9 years ago
1
Fail to build the webrtc2sip code

#1 GoogleCodeExporter closed 9 years ago
3