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webrtc2sip
Automatically exported from code.google.com/p/webrtc2sip
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webrtc2sip crash on call hold from chrome browser
#177
GoogleCodeExporter
opened
9 years ago
0
Assertion Failure while running voice call
#176
GoogleCodeExporter
opened
9 years ago
0
stun server installation for webrtc2sip
#175
GoogleCodeExporter
opened
9 years ago
1
Webrtc2sip on Ubuntu 14.04 not working
#174
GoogleCodeExporter
opened
9 years ago
0
webrtc2sip certificate issues
#173
GoogleCodeExporter
opened
9 years ago
2
webrtc2sip + sipml5 : MSG: WS handshaking not done yet
#172
GoogleCodeExporter
opened
9 years ago
0
webrtc2sip build error - ‘strnicmp’ was not declared in this scope
#171
GoogleCodeExporter
closed
9 years ago
1
webrtc2sip build error - ‘strnicmp’ was not declared in this scope
#170
GoogleCodeExporter
closed
9 years ago
1
crash when closing connection
#169
GoogleCodeExporter
closed
9 years ago
1
webrtc2sip fails to build: undefined references
#168
GoogleCodeExporter
closed
9 years ago
3
no Sound, getting errors on commandline
#167
GoogleCodeExporter
opened
9 years ago
1
ACK to 401 Sent to Wrong IP
#166
GoogleCodeExporter
opened
9 years ago
1
webrtc2sip: ../src/pj/os_core_unix.c:674: pj_thread_this: Assertion `!"Calling pjlib from unknown/external thread. You must " "register external threads with pj_thread_register() " "before calling any pjlib functions."' failed.
#165
GoogleCodeExporter
closed
9 years ago
2
webtc2sip crash on re-INVITE
#164
GoogleCodeExporter
opened
9 years ago
0
Current test for "have libs" in configure.ac (line 114) expects 13 "yes", but 14 are required
#163
GoogleCodeExporter
closed
9 years ago
1
Configure --with-ffmpeg fails
#162
GoogleCodeExporter
opened
9 years ago
1
exception occurs in webrtc2sip when caller hangs up.
#161
GoogleCodeExporter
closed
9 years ago
1
RTCweb breaker includes invalid ice-pwd attribute in SDP
#160
GoogleCodeExporter
closed
9 years ago
3
DTLS handshake failed [error:14102418:SSL routines:DTLS1_READ_BYTES:tlsv1 alert unknown ca]
#159
GoogleCodeExporter
opened
9 years ago
1
SRTP establishing problem when webbreaker=true
#158
GoogleCodeExporter
opened
9 years ago
0
No Ciphers Available when attempting WSS between SipML 1.3 / 1.5 and WebRTC 2.6.0
#157
GoogleCodeExporter
opened
9 years ago
6
configure searches for doubango 2.0.1089, only 2.0.1086 available
#156
GoogleCodeExporter
closed
9 years ago
2
webrtc2sip configure failed
#155
GoogleCodeExporter
closed
9 years ago
2
webrtc2sip crash
#154
GoogleCodeExporter
closed
9 years ago
1
Call not being dropped after error message received on Firefox 26
#153
GoogleCodeExporter
opened
9 years ago
0
webrtc2sip is not sending 200 OK to SIP MESSAGE sender for a delivered SIP MESSAGE where both users are served by webrtc2sip gw supported by a SIP server in the back-end
#152
GoogleCodeExporter
opened
9 years ago
3
Ability to configure the SIP Proxy (IP and port) from webrtc2sip server configuration file
#151
GoogleCodeExporter
opened
9 years ago
0
random crashs
#150
GoogleCodeExporter
closed
9 years ago
1
No Media (audio) When The Call Gets Connected to Voice Mail on a SIP Server
#149
GoogleCodeExporter
opened
9 years ago
0
Disconnected/Unautorized running siplm5 live demo with sip2sip recommended
#148
GoogleCodeExporter
opened
9 years ago
1
SDP Parse Error
#147
GoogleCodeExporter
opened
9 years ago
0
Private extensions
#146
GoogleCodeExporter
closed
9 years ago
1
MSG: Failed to start the RTP/RTCP transport
#145
GoogleCodeExporter
opened
9 years ago
0
When will be a new release?
#144
GoogleCodeExporter
closed
9 years ago
1
no audio when using mobile connection
#143
GoogleCodeExporter
opened
9 years ago
0
Stun? or webRTC
#142
GoogleCodeExporter
opened
9 years ago
0
Video V8 freez
#141
GoogleCodeExporter
opened
9 years ago
0
webrtc2sip: double free or corruption (fasttop)
#140
GoogleCodeExporter
closed
9 years ago
4
Connection UDP
#139
GoogleCodeExporter
opened
9 years ago
0
WebRTC2SIP error in code
#138
GoogleCodeExporter
closed
9 years ago
1
Using stun server leads to termination of call abruptly (WebRTC Gateway)
#137
GoogleCodeExporter
opened
9 years ago
1
webrtc2sip 2.6.0 not started
#136
GoogleCodeExporter
opened
9 years ago
4
duplicate symbol when building in MAC OSX 10.8.5
#135
GoogleCodeExporter
closed
9 years ago
1
doubango and webrtc2sip configure.ac mismatch in svn r118 causes build to fail
#134
GoogleCodeExporter
closed
9 years ago
1
Allow setting "TSIP_TRANSPORT_STREAM_PEER_TIMEOUT" in config.xml
#133
GoogleCodeExporter
opened
9 years ago
0
A call with a SIP endpoint over TCP results in a call disconnection when the TCP SIP connection closes.
#132
GoogleCodeExporter
opened
9 years ago
1
Incorrect routing by webrtc2sip when two Route headers are present
#131
GoogleCodeExporter
opened
9 years ago
3
force rtcp passthrough
#130
GoogleCodeExporter
opened
9 years ago
0
cannot hear anything on both the ends
#129
GoogleCodeExporter
closed
9 years ago
2
DTLS-SRTP negotiation does not happen sometimes
#128
GoogleCodeExporter
opened
9 years ago
2
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