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webrtc2sip
Automatically exported from code.google.com/p/webrtc2sip
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JitterBuffer enbale/disable option is useless
#78
GoogleCodeExporter
closed
9 years ago
1
Delay in H.264
#77
GoogleCodeExporter
closed
9 years ago
1
One way audio from sipml5 to freeswich
#76
GoogleCodeExporter
opened
9 years ago
1
180 Ringing and early media sent to early
#75
GoogleCodeExporter
opened
9 years ago
2
Correct digest authentication call flow for authentication of INVITEs at the network
#74
GoogleCodeExporter
opened
9 years ago
5
Fails to connect when UA is behind a symetric NAT
#73
GoogleCodeExporter
opened
9 years ago
0
Adds supports for Reduced-Size Real-Time Transport Control Protocol (rfc 5506)
#72
GoogleCodeExporter
opened
9 years ago
0
Adds support for DTMF
#71
GoogleCodeExporter
closed
9 years ago
1
[Doubango] Closing websocket issue
#70
GoogleCodeExporter
closed
9 years ago
1
Call from Android to Chrome using the RTCWeb Breaker = no video
#69
GoogleCodeExporter
closed
9 years ago
2
Adds support for mysql connector
#68
GoogleCodeExporter
opened
9 years ago
0
When RTCWeb Breaker is enabled - SIP INVITE authentication fails
#67
GoogleCodeExporter
opened
9 years ago
4
No audio from webrtc2sip to Chrome when attaching a legacy device
#66
GoogleCodeExporter
opened
9 years ago
7
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
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