havfo WEBRTC-to-SIP issues

havfo / WEBRTC-to-SIP

Setup for a WEBRTC client and Kamailio server to call SIP clients

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issues

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use kamailio as relay

#43 gentoosys opened 1 month ago
0
Audio problem if rtpengine_manage() is called when there is no SDP body.

#42 antonio-couto opened 2 months ago
1
Audio issues from Webrtc to SIP

#41 Voipdevel closed 4 months ago
4
E: Package 'ngcp-keyring' has no installation candidate

#40 Kuanlin-Chen opened 7 months ago
0
why use turn server, can I only use rtpengine？

#39 Mr-houzi closed 8 months ago
3
Dockerfile ?

#38 rgl1234 opened 1 year ago
0
SIP to WebRTC calls only one way audio

#37 m1ggy closed 2 years ago
5
webrtc solution for sip protocol

#36 vitorpc4 closed 2 years ago
0
Kamailio + RTPEngine + TURN server behind NAT

#35 yboujraf closed 2 years ago
5
Update GPG keys for SIPWise repo

#34 tomd-tc closed 3 years ago
0
Sip WIse installation in ubuntu 18.04

#33 sudipta1411 closed 3 years ago
1
could not login sipml5 demo to kamailio via websocket.

#32 voipmanvn closed 3 years ago
3
Testtest

#31 DaHermann closed 3 years ago
0
NO_PUBKEY Error in debian 10

#30 vigikaran closed 3 years ago
1
Update tls.cfg

#29 vaclavrak closed 3 years ago
1
Integration with Asterisk

#28 27042017 closed 3 years ago
1
No audio from SIP to WS

#27 chiragd closed 2 years ago
2
When lets encrypt permission denied

#26 thadeu closed 3 years ago
0
Add dispatcher

#25 furek1 closed 4 years ago
4
E: Unable to correct problems, you have held broken packages.

#24 webrtcccccc closed 3 years ago
2
Error!! Unmet build dependencies : libbcg729-dev

#23 Diamond555 closed 5 years ago
0
Parallel call forking case

#22 korayvt closed 5 years ago
6
SIP Client with WebRTC SDP

#21 ForGuru closed 5 years ago
1
Build doesn't work

#20 mSys-mislav closed 5 years ago
2
reinvite after iceRestart does not work

#19 bearinld004 closed 6 years ago
3
at what condition is turn actually used?

#18 andrewvmail closed 6 years ago
2
Docket compose

#17 virus2016 opened 6 years ago
0
iptables.sh issue

#16 salihy closed 6 years ago
4
can the RTPEngine component replaced by Kurento?

#15 deltapath-eric closed 6 years ago
2
can i run all of in same server?

#14 bunalng closed 5 years ago
5
chat error

#13 dipenpatel235 closed 6 years ago
3
file flavors/no_ngcp not found

#12 kossivi1984 closed 6 years ago
6
How does this work without transcoding?

#11 whitebook closed 6 years ago
1
Handling Late-Offer Re-invites

#10 sudermanjr closed 6 years ago
2
getting error at chat conversation

#9 prashantvidja closed 7 years ago
2
how to register multiple register at WEBRTC

#8 prashantvidja closed 7 years ago
2
need to consulting to fix briding to work with asterisk behing the nat

#7 avbdr closed 7 years ago
4
Internal / external network setup

#6 BertrandTotti closed 6 years ago
4
sip.js and sdp error 488

#5 balabanferhat closed 6 years ago
7
SDP offer/answer issue

#4 faizann closed 6 years ago
3
b/f:use NAT handling for 183->180 replies, too

#3 sanchi closed 7 years ago
0
b/f: use rtpengine_manage in all cases when rewriting SDP

#2 sanchi closed 7 years ago
0
can i contract you to set this up for me?

#1 bhakimi closed 7 years ago
0