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Does the current SIP component support SIP interworking with third-party platforms? #184

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sunnytzh opened this issue Sep 25, 2024 · 1 comment
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@sunnytzh
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1、Does the current SIP component support interworking between SIP and other platforms? Why SIP components in the code only audio processing, no video processing. Sip handshake after success, what should be done next, feel not introduced, is not this part of the function is not perfect.
2、For now, create SIP participant is successful, but there is nothing to say about how it should be handled next. Can We give some suggestions after the SIP handshake is successful.

@dennwc dennwc self-assigned this Sep 25, 2024
@dennwc dennwc added the question Further information is requested label Sep 25, 2024
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dennwc commented Sep 25, 2024

Does the current SIP component support interworking between SIP and other platforms?

If you are asking if SIP server can be used with other generic SIP services (as opposed to SIP trunks), then yes, it should work as long as the other service uses INVITE and not REGISTER.

If you are asking about any other non-SIP applications, then it's easier to integrate LiveKit with that platform directly instead of using SIP as an intermediate protocol.

Why SIP components in the code only audio processing, no video processing.

We do not support video for SIP yet, because LiveKit rooms may have multiple participants, while SIP typically supports just one. Thus, it will require an active speaker detection or video mixing implemented in SIP. We may add support in the future, but it wasn't the priority at the moment.

Sip handshake after success, what should be done next, feel not introduced, is not this part of the function is not perfect.

There's an (outdated) PR here, that introduced basic support for video: #35 . You may try doing something similar.

For now, create SIP participant is successful, but there is nothing to say about how it should be handled next. Can We give some suggestions after the SIP handshake is successful.

When LiveKit SIP is configure correctly (Inbound Trunk + Dispatch Rule is created), the call will end up in a LiveKit room selected by a Dispatch Rule. You can then join that room from Web using LiveKit Meet (or any other LiveKit client or SDK).

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