pjmedia/pjsua: add on-demand stream keep-alive API - #15
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darshan-verkada merged 2 commits intoJun 17, 2026
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Add pjmedia_stream_send_keep_alive(), pjsua_call_send_stream_keep_alive() and pjsua2 Call::sendStreamKeepAlive() to send an empty RTP + RTCP keep-alive on demand, independent of PJMEDIA_STREAM_ENABLE_KA (whose auto-KA only fires from within put_frame()). This lets an application keep the media path / NAT binding alive when the local sound device has been disconnected from the stream (parked audio) but the SIP/media session must stay up, without reaching into the raw pjmedia stream port and calling put_frame() from an app thread. The pjsua-lib entry point sends under PJSUA_LOCK and revalidates the stream, so it is mutually exclusive with pjsua_media_channel_deinit() and cannot use-after-free the stream during call teardown. Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
MattMacGregor
requested changes
Jun 15, 2026
Per review feedback: since PJMEDIA_STREAM_ENABLE_KA=1 in Verkada builds, send_keep_alive_packet is always compiled in. Replace the duplicated logic in pjmedia_stream_send_keep_alive with a direct call-through; return PJ_ENOTSUP as a fallback for builds with KA disabled. Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
MattMacGregor
approved these changes
Jun 16, 2026
darshan-verkada
merged commit Jun 17, 2026
b3f126f
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intercom/release-2-15
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Summary
pjmedia_stream_send_keep_alive()— a build-flag-independent public API that sends an empty RTP packet + RTCP to keep the media path / NAT binding alive whenput_frame()is not driving the stream (e.g. local audio device disconnected but SIP session must stay up).pjsua_call_send_stream_keep_alive(call_id, med_idx)— a pjsua-level wrapper that acquiresPJSUA_LOCKbefore touching the stream, making it mutually exclusive withpjmedia_channel_deinit(). This prevents a use-after-free where the caller's thread could race stream teardown.Call::sendStreamKeepAlive(med_idx)pjsua2 C++ binding.Test plan
Make intercom call using command, make sure it switches to
livekit. check SIP audio threads are closed, but call is still active for 2 minutes.🤖 Generated with Claude Code