Giters
livekit
/
sip
SIP to WebRTC bridge for LiveKit
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Stargazers:
73
Watchers:
15
Issues:
31
Forks:
19
livekit/sip Issues
Add early SIP dialog state
Updated
2 months ago
Comments count
1
TCP/TLS support for SIP signaling
Updated
2 months ago
Comments count
7
Codec negotiation issue
Closed
2 months ago
Comments count
2
In-dialog SIP INVITE breaks the call session
Updated
2 months ago
Comments count
2
Support for the AMR-WB codec for HD Voice on T-Mobile
Updated
2 months ago
Comments count
1
Call transfer via SIP REFER
Updated
2 months ago
Comments count
6
Templated Metadata
Updated
2 months ago
Comments count
3
Run with Livekit-client-sdk-js
Closed
3 months ago
Comments count
2
Video support
Closed
7 months ago
Comments count
15
publish error
Updated
4 months ago
Comments count
3
running test client main.go get this log
Updated
6 months ago
outbound error
Updated
6 months ago
Comments count
6
Unhandled response with Telnyx outbound
Updated
6 months ago
Comments count
4
Outbound call client unable to respond
Updated
6 months ago
Comments count
10
Can you create a video explaining how to use it and post it on YouTube?
Closed
7 months ago
Comments count
6
No SIP Trunk or Dispatch Rules matched.But have it
Closed
7 months ago
Comments count
2
demo of inboud rule
Updated
7 months ago
Comments count
1
which profile should i use for video call
Closed
7 months ago
Comments count
1
Dialing up to join the association reported an error, seeking help
Updated
7 months ago
Can you provide the client code for testing?
Closed
7 months ago
Comments count
2
how to make call out
Closed
7 months ago
Comments count
4
create sip trunk
Closed
7 months ago
Comments count
5
livekit/sip docker image not found
Closed
7 months ago
Comments count
3
do we support video mcu mode?
Closed
7 months ago
Comments count
1