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webrtc2sip
Automatically exported from code.google.com/p/webrtc2sip
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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
Cannot hear audio video.
#127
GoogleCodeExporter
closed
9 years ago
1
issue for incoming calls, audio and puch notification.
#126
GoogleCodeExporter
closed
9 years ago
6
Webrtc2sip set up and runs but won't allow connection from sipml5
#125
GoogleCodeExporter
opened
9 years ago
1
cannot find audio producer for left leg.and no audio after call establishment
#124
GoogleCodeExporter
opened
9 years ago
0
No audio and video after call establish after 25 seconds
#123
GoogleCodeExporter
opened
9 years ago
0
webrtc2sip suddenly down.
#122
GoogleCodeExporter
closed
9 years ago
13
Make refCount inc/dec thread-safe
#121
GoogleCodeExporter
opened
9 years ago
0
segfault error 4 in libtinyDAV.so.0.0.0 when more than 20 calls are fired within a second
#120
GoogleCodeExporter
opened
9 years ago
0
console output reports use of proxy: ns313841.ovh.net:13062
#119
GoogleCodeExporter
closed
9 years ago
1
BYE not forwarded from WebRT2Sip Gateway to Outbound Proxy
#118
GoogleCodeExporter
opened
9 years ago
0
SIP Reinvite not working when using sipml5 and mobicents
#117
GoogleCodeExporter
opened
9 years ago
3
Address already in use
#116
GoogleCodeExporter
closed
9 years ago
1
SetRemoteDescription failed: Called with a SDP without crypto enabled.
#115
GoogleCodeExporter
closed
9 years ago
4
webrtc2Sip too many files open
#114
GoogleCodeExporter
opened
9 years ago
1
Getting issue with G729 and
#113
GoogleCodeExporter
closed
9 years ago
1
ERROR: function: "tnet_sockfd_sendto()"
#112
GoogleCodeExporter
opened
9 years ago
1
DTLS handshake error
#111
GoogleCodeExporter
closed
9 years ago
2
Allow setting max up and down bandwidth
#110
GoogleCodeExporter
opened
9 years ago
0
CSeq sequence apparently out of standard
#109
GoogleCodeExporter
closed
9 years ago
6
Receiving on console ***ERROR: function: "tsip_message_parse()" line: "227" MSG: Failed to parse SIP message
#108
GoogleCodeExporter
closed
9 years ago
2
No audio and video after call establish.
#107
GoogleCodeExporter
opened
9 years ago
0
webrtc2sip on windows
#106
GoogleCodeExporter
opened
9 years ago
2
Doubango 283
#105
GoogleCodeExporter
opened
9 years ago
0
Doubango 282
#104
GoogleCodeExporter
opened
9 years ago
0
Skype VS Webrtc2sip - firewall issues
#103
GoogleCodeExporter
closed
9 years ago
1
error when installing webrtc2sip task-debug.h erreur fatale
#102
GoogleCodeExporter
closed
9 years ago
3
Fail to install: mp_object.h:23:23: fatal error: tsk_debug.h: No such file or directory
#101
GoogleCodeExporter
closed
9 years ago
7
webrtc2sip does not compile using latest build
#100
GoogleCodeExporter
closed
9 years ago
9
How to configure webrtc2sip gateway to connect to IMS core
#99
GoogleCodeExporter
opened
9 years ago
1
wrong RTP/RTCP Manager binding
#98
GoogleCodeExporter
closed
9 years ago
2
webrtc2sip gets into stuck state under load.
#97
GoogleCodeExporter
opened
9 years ago
8
CPU 100% used when webrtc2sip run
#96
GoogleCodeExporter
closed
9 years ago
2
service webrtc2sip status exception
#95
GoogleCodeExporter
closed
9 years ago
2
webrtc2sip segfaults whn using a hardcoded config path and setting config path on the command line
#94
GoogleCodeExporter
closed
9 years ago
2
Can't make call through webrtc2sip (Forbidden)
#93
GoogleCodeExporter
closed
9 years ago
2
Allow setting "disbale_stun_lookup" using the xml config file
#92
GoogleCodeExporter
closed
9 years ago
1
webrtc2sip initial service setup
#91
GoogleCodeExporter
closed
9 years ago
5
webrtc2sip b2bua failed to do digest auth for INVITE
#90
GoogleCodeExporter
closed
9 years ago
5
Calling PSTN create resampler (opus->g711)
#89
GoogleCodeExporter
opened
9 years ago
1
Set codec priorities as they appear in the config list
#88
GoogleCodeExporter
closed
9 years ago
1
Video call to audio endpoint - result one way audio
#87
GoogleCodeExporter
opened
9 years ago
3
webrtc2sip+asterisk+sipml5
#86
GoogleCodeExporter
closed
9 years ago
2
Chrome to Chrome webRTC Calling ... Unexpected Media Issue with various Breaker configuration
#85
GoogleCodeExporter
closed
9 years ago
4
RTP not passing on to the Google Chrome
#84
GoogleCodeExporter
opened
9 years ago
4
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