Feature Request: RFC 7198 RTP Stream Duplication for Broadcast Resilience

[broadcasty4201]

Request for RFC 7198 (Duplicating RTP Streams) support in baresip, to enable resilient SIP audio connections to broadcast infrastructure that already supports this standard.

We are using baresip as a SIP contribution codec connecting to professional broadcast infrastructure — specifically Tieline Gateway units used for radio outside broadcasts and studio contributions. These Tieline Gateways are fully RFC 7198 compliant and advertise dual stream support in their SDP.

For broadcast contribution use, packet loss resilience without increased latency is critical. RFC 7198 achieves this by sending duplicate RTP streams. The receiver reconstructs from whichever copy arrives intact. This is now standard in the broadcast SIP codec market (Vortex CallMe, Tieline, AETA all support it), but we have no way to take advantage of it from baresip.

Is there an existing or planned mechanism to hook the RTP send path from a module, or would this require changes to stream.c?
Is there any existing work on RFC 7198 or related stream duplication in any branch?
Would a contribution implementing this be welcome as a pull request?

Thank you!