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DoubangoTelecom
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webrtc2sip
Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network
https://doubango.org
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libvpx.so.1 not found
#199
llKoull
opened
4 years ago
0
Operation maintenance
#198
pepsi7959
opened
4 years ago
0
Integration with freepbx or fusionpbx
#197
ugintl
opened
4 years ago
1
set RTP/SRTP with configuration
#196
pepsi7959
closed
5 years ago
1
Redirect call from webapp to sip phone with 302 Moved Temporarily message
#195
wieczorekmichal1
opened
5 years ago
0
How to compile webrtc2sip correctly?
#194
XDhughie
opened
5 years ago
1
Got a problem with calls
#193
praeitor
opened
8 years ago
1
Not respecting original User-Agent on INVITE and REGISTER
#192
scgm11
closed
7 years ago
0
High CPU during calls also with same codecs
#191
epasqualotto
opened
8 years ago
0
Using WSS with WebRTC and Letsencrypt (Let's Encrypt) via sipML = "no shared cipher"
#190
chrischarles2002
opened
8 years ago
1
Installed webrtc2sip but can't use the command
#189
Ganasci
opened
8 years ago
0
tinywrap folder should be external from doubango project
#188
DoubangoTelecom
opened
8 years ago
0
No Audio if call is accepted after 30 seconds
#187
vinodpandey23
opened
8 years ago
2
The git directory miss the tinywrap folder and it's contents and can therefore not compile
#186
legobit
opened
8 years ago
4
Webrtc2sip crashing randomly in SIPML5 scenario
#185
GoogleCodeExporter
opened
9 years ago
1
how to make sip client direct to gateway ?
#184
GoogleCodeExporter
opened
9 years ago
1
Webrtc2sip crash when media coder is enabled.
#183
GoogleCodeExporter
opened
9 years ago
0
183/early media not forwarded to browser by webrtc2sip
#182
GoogleCodeExporter
opened
9 years ago
0
Unable to start webrtc2sip using init.d script
#181
GoogleCodeExporter
opened
9 years ago
0
asterisk 11.2.1+sipml5+webrtc2sip - No audio in chrome, audio available in firefox
#180
GoogleCodeExporter
opened
9 years ago
1
Problem in BYE send from Webrtc2sip (Version 2.6.0)
#179
GoogleCodeExporter
opened
9 years ago
1
Webrtc2sip Installing but net/if_dl.h / net/if_types.h not found in doubango compiling
#178
GoogleCodeExporter
opened
9 years ago
0
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
6
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
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