baresip, multicast, Raspberry Pi, and Innomaker RPI HiFi AMP HAT #3022
Replies: 3 comments 2 replies
-
We shortly moved multicast module to baresip-apps.
Resampling, more concretely up-sampling need to be done. This can be done by module
Note, that only divisible input/output sample rates are supported! In this case (16000 -> 48000 , 8000 -> 48000) should work. |
Beta Was this translation helpful? Give feedback.
-
Thanks for the reply. I appreciate the heads up about the move to baresip-apps. I'll make sure to include the module from there the next time I compile. I am using G.722 for the audio multicasts, and, as I said, it works nicely on all of my the baresip instances. Only the one running the Innomaker RPI HiFi AMP HAT fails, and only for multicast, not SIP pages. The audio is already formatted to 48000 Hz, but I also have the auresamp.so module available and enabled. If there some debugging or other information that I could share that might be helpful? Thanks again! |
Beta Was this translation helpful? Give feedback.
-
So, it looks like baresip is converting incoming SIP pages from 16000 Hz to 48000 Hz, but not incoming multicast audio. ... multicast receiver: start addr=224.0.36.10:10010 prio=2 enabled=1 state=receiving alsa: reset: srate=16000, ch=1, num_frames=320, pcmfmt=S16_LE alsa: playback started (default) multicast receiver: start addr=224.0.36.10:10010 prio=2 enabled=1 state=running auresamp: resample decoder 16000/1 --> 16000/1 multicast receiver: EOS addr=224.0.36.10:10010 prio=2 enabled=1 state=running alsa: stopping playback thread (default) ua: no call to resume ua: sipsess connect via UDP 172.16.0.6:5060 --> 172.16.5.141:35312 7341@172.16.0.6: selected for 7341-0x55acf20bb0 ua: using connection-address 172.16.0.6 of SDP offer call: alloc with params laddr=172.16.5.141, af=AF_INET, use_rtp=1 call: use_video=0 call: answering call on line 1 from sip:7230@172.16.0.6 with 200 audio: peer changed ptime_tx 40ms -> 20ms stream: update 'audio' stream: disable audio RTP receiver stream: disable audio RTP sender stream: enable audio RTP receiver stream: enable audio RTP sender stream: audio: starting RTCP with remote 172.16.0.6:15151 audio: update audio: Set audio decoder: G722 16000Hz 1ch audio: start alsa: reset: srate=48000, ch=1, num_frames=1920, pcmfmt=S16_LE alsa: playback started (default) audio: player started with sample format S16LE audio: Set audio encoder: G722 16000Hz 1ch audio: start aubridge: created device 'dummy' audio: source started with sample format S16LE audio tx pipeline: aubridge ---> aubuf ---> auconv ---> auresamp ---> G722 audio rx pipeline: alsa <--- aubuf <--- auconv <--- auresamp <--- G722 call: stream start (active=1) audio: Set audio decoder: G722 16000Hz 1ch audio: start stream: enable audio RTP receiver stream: enable audio RTP sender 7341@172.16.0.6: Call established: sip:7230@172.16.0.6 stream: incoming rtp for 'audio' established, receiving from 172.16.0.6:15150 auresamp: resample decoder 16000/1 --> 48000/1 audio_recv: create audio buffer [20 - 160 ms] [1920 - 15360 bytes] sip:7230@172.16.0.6: session closed: Connection reset by peer [104] stream: disable audio RTP receiver ... Can someone confirm that is what these logs say? And, if so, any suggestions as to how to ask baresip to convert for multicast audio? Thank you! |
Beta Was this translation helpful? Give feedback.
-
I'm having an odd problem with using baresip to play multicast audio with a Raspberry Pi and the Innomaker RPI HiFi AMP HAT (https://www.inno-maker.com/product/hifi-amp-hat/ ).
I have build about 30 baresip PAs using both old Mac Minis and Raspberrry Pis. They are configured as SIP endpoints and to accept multicast audio and work fine.
My existing Raspberry Pi PAs use the built-in audio (for the Raspberry Pi 4s) or a USB audio interface (for the Raspberry Pi 5s).
My goal now it so build some Raspberry PI 4s with the Innomaker RPI HiFi AMP HAT so I can connect them directly to loudspeakers.
I have a prototype working fine at a SIP endpoint, but, while it receives the multicast audio - it makes no sound. That is, it plays sound from thee ALSA aplay tool and from baresip when I page its extension, but (while I can see that it receives the incoming multicast audio from the console) it makes no sound when it gets multicast audio.
I did have to configure baresip to use use the dummy interface for the mic and lock it to 48000 hz to get it to play SIP pages.
~/.baresip/config
I'm wondering if there is something similar that I am missing in order to make the Innomaker amp work with multicast audio.
Thanks everyone. I appreciate any ideas or suggestions you may have!
Beta Was this translation helpful? Give feedback.
All reactions