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SIP to WebRTC bridge for LiveKit
Hello, thanks for your great work on this sytem. I am wondering if we can add AMR-WB codec support. This seems to be the codec used by T-mobile for HD...
It _appears_ that the SIP adapter/service is currently hard coded to use UDP for SIP signaling. However, it appears the upstream library has support for TCP and TLS. https://github.com/livekit/sip/blob/1ce2ec84ebf08f23d97febddb507f63d4cc21c6d/pkg/sip/server.go#L172 Would...
I wasn't able to find any documentation/source code reference that enables me to transfer an ongoing call (e.g., in the case of a human handover). Is transferring a call supported...
The metadata that is included with a SIP participant can be defined in the configuration, but appears to be static. Would there be any interest in (or aversion to) allowing...
dear author, when i try to call into livekit server via sip , sip message all is well ,but there have no rtp, and then i found some error at...
 The web end can display SIP membership, but there is no audio available
dear author, when i user outbound api to make call, it worked.but when i hangup the sip endpoint ,the livekit room still exist the sip endpoint ,the logs like below,...
Hi, I'm attempting to connect over a Telnyx trunk to a my dev server. The outgoing call works fine, but once connected I am receiving `Unhandled sip response. UnhandledResponseHandler handler...
Hello, I need your help. Thank you very much. I am unable to respond during the outbound call test, as shown in the video.    https://github.com/livekit/sip/assets/110219509/2f42c327-2ad9-41de-a143-fecfec3ebb8f
hi author， i added direct rule and bind trunk ，sometimes the route it worked ，but sometimes it cannot work well.could u give some real demo about create trunk and rule(direct)...