The rtp-sink module creates a PipeWire sink that sends audio RTP packets.
libpipewire-module-rtp-sink
Options specific to the behavior of this module
source.ip =<str>: source IP address, default "0.0.0.0"destination.ip =<str>: destination IP address, default "224.0.0.56"destination.port =<int>: destination port, default random between 46000 and 47024local.ifname = <str>: interface name to usenet.mtu = <int>: MTU to use, default 1280net.ttl = <int>: TTL to use, default 1net.loop = <bool>: loopback multicast, default falsesess.min-ptime = <float>: minimum packet time in milliseconds, default 2sess.max-ptime = <float>: maximum packet time in milliseconds, default 20sess.name = <str>: a session namertp.ptime = <float>: size of the packets in milliseconds, default up to MTU but between sess.min-ptime and sess.max-ptimertp.framecount = <int>: number of samples per packet, default up to MTU but between sess.min-ptime and sess.max-ptimesess.latency.msec = <float>: target node latency in milliseconds, default as rtp.ptimesess.ts-offset = <int>: an offset to apply to the timestamp, default -1 = random offsetsess.ts-refclk = <string>: the name of a reference clocksess.media = <string>: the media type audio|midi|opus, default audiostream.props = {}: properties to be passed to the streamaes67.driver-group = <string>: for AES67 streams, can be specified in order to allow the sink to be driven by a different node than the PTP driver.Options with well-known behavior: