Initial implementation of Hold/Resume call

[Kaian]

We've been exploring the idea of enabling real attended call transfers from LiveKit rooms, with the first step being support for call holding. When connecting LiveKit Trunks from a PBX, this allows music-on-hold configured in the PBX to be played to the participant.

This PR introduces the initial implementation of call hold and unhold functionality in the LiveKit SIP component. It allows the LiveKit Server to put SIP calls on hold and resume them (unhold), by adding the necessary handling of re-INVITE messages for hold/unhold.

Further PRs will be required in other LiveKit projects to fully implement this functionality across the platform (including livekit, livekit-sdks, livekit-protocol and so on).

We’d love to hear if the team is interested in this functionality and any feedback on the approach to summit additional Pull Requests.

[CLAassistant]

All committers have signed the CLA.

[codecov]

Codecov Report

❌ Patch coverage is 10.10101% with 356 lines in your changes missing coverage. Please review.
✅ Project coverage is 59.97%. Comparing base (0460b40) to head (04d94e4).
⚠️ Report is 131 commits behind head on main.

Files with missing lines	Patch %	Lines
pkg/sip/inbound.go	2.56%	150 Missing and 2 partials ⚠️
pkg/sip/outbound.go	2.56%	152 Missing ⚠️
pkg/sip/service.go	0.00%	50 Missing ⚠️
pkg/service/service.go	88.88%	2 Missing ⚠️

Additional details and impacted files

@@            Coverage Diff             @@
##             main     #443      +/-   ##
==========================================
- Coverage   65.25%   59.97%   -5.29%     
==========================================
  Files          51       32      -19     
  Lines        6588     7105     +517     
==========================================
- Hits         4299     4261      -38     
- Misses       1915     2462     +547     
- Partials      374      382       +8     

☔ View full report in Codecov by Sentry.
📢 Have feedback on the report? Share it here.

🚀 New features to boost your workflow:

• ❄️ Test Analytics: Detect flaky tests, report on failures, and find test suite problems.

[dennwc]

Thank you a lot for contributing the hold/resume functionality! The idea and implementation makes perfect sense. New internal API is the most straightforward way to achieve this.

What I'm wondering about is, would it make sense to hold/unhold automatically when SIP participant looses all audio track subscriptions? In this case, if a person/agent that is in the room suddenly drops, we could indicate a hold for the SIP participant instead of just relaying silence.

This way, there will be no extra API required - existing participant admin API can be used to unsubscribe SIP from existing tracks, move it to another empty room or disconnecting the other peer from the room.

What do you think about this approach?

[Kaian]

Hi!

Thanks for the feedback!

We're currently testing this feature in the following scenario: An incoming call is dispatched to a room where an agent can be requested to perform a transfer, first checking whether the destination is available. The agent places the first participant on hold and creates a second participant in the same room by making an outbound call. When this second participant answers, their availability is verified, and based on the response, we remove the second participant from the room and perform a cold transfer from the first participant (or not). This approach has some advantages, such as providing a full transcription or recording of the entire process while only using a single room.

I have little to no experience with LiveKit, so I cannot say what the best approach would be. However, I’m concerned that some providers may not accept in-dialog re-INVITEs the way PBXs do, so making this behavior implicit could cause future issues in other scenarios.

Of course, your opinion is what matters the most, and if you decide to implement it differently, we’ll be more than happy to adapt how our agent works!

Best regards,