How can I make sure the following options are added to the client offer SDP generated by GStreamer webrtcbin (Android), please?
minptime=10;useinbandfec=1
Background:
===========
Pipeline generated for receive only case by parsing:
#define RECV_PIPELINE "webrtcbin name=recv bundle-policy=max-bundle recv.src_0 ! queue ! rtpjitterbuffer do-lost=TRUE ! rtpopusdepay ! opusdec use-inband-fec=TRUE plc=TRUE ! audioconvert ! audioresample ! volume name=vol ! level message=TRUE ! openslessink stream-type=voice"
then adding a transceiver:
GstCaps *caps = gst_caps_from_string("application/x-rtp,media=audio,encoding-name=OPUS,payload=111,clock-rate=48000");
GstPromise *promise = gst_promise_new();
g_signal_emit_by_name(webrtcbin, "add-transceiver",
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, caps, promise);
gst_promise_interrupt(promise);
gst_promise_unref(promise);
gst_caps_unref(caps);
Generates offer SDP:
v=0
o=- 574272700240657154 0 IN IP4 0.0.0.0
s=-
t=0 0
a=ice-options:trickle
a=group:BUNDLE audio0
m=audio 9 UDP/TLS/RTP/SAVPF 111
c=IN IP4 0.0.0.0
a=setup:actpass
a=ice-ufrag:blah
a=ice-pwd:blah
a=rtcp-mux
a=rtcp-rsize
a=recvonly
a=rtpmap:111 OPUS/48000
a=rtcp-fb:111 nack pli
a=mid:audio0
a=fingerprint:sha-256 blah
When libwebrtc is used instead, the SDP includes:
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
I'm trying to optimise behaviour seen under lossy conditions; at the moment my webrtcbin version seems to be suffering from more noticeable interruptions than libwebrtc & has some additional latency. The OPUS options seem like they might be a contributing factor...?
I see plenty of log lines like this:
gst_audio_base_sink_get_alignment:<openslessink0> Unexpected discontinuity in audio timestamps of -0:00:00.226541666, resyncing
Apologies: more than one question in here!
thanks very much for any insight
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