rpm-arch
commented
1 year ago (sorry accidental close - issue resolved with workaround but may still be considered bug)
Figured it out - if there is no input level (in my case, not porting mic input over the rdp session), and if echo cancellation is enabled (it is by default), then there is no RTP leg established outbound. I believe this error is related:
2023-01-09 19:24:51:784 [AppRun.wrapped/bctbx] WARNING Getting reference signal but no echo to synchronize on.
Not establishing the RTP leg outbound results in a couple issues:
- If any codecs are enabled other than the codec received on inbound RTP (in my case, PCMA - payload type 8), then ortp/mediastreamer drops the packets as unknown payload:
2023-01-09 18:46:49:625 [AppRun.wrapped/ortp] WARNING Receiving packet with unknown payload type 8. 2023-01-09 18:46:49:625 [AppRun.wrapped/mediastreamer] WARNING Discarding packet with unknown payload type 8
- Even if the payload is accepted by limiting the codecs enabled on the account to only that one (resolving the log messages above), the sound represented in the linphone interface is still not heard nor represented in the gnome sound settings level (as per the second screenshot in my original post). Not sure exactly why this happens.
So, the workaround is simply disabling echo cancellation. Still, I am not sure why it is needed to have an outbound RTP payload to negotiate the codec or moreover for the sound which is shown in the app to be shared to pulseaudio.
I have confirmed this issue still exists on the latest appimage (5.0.5) using an x86 instance instead of aarch64.
Context
I am using linphone as it seems to be the only maintained softphone available for current Ubuntu on ARM. Purpose is to create a SIP calling demonstration for my company.
General information
For remote access using: https://github.com/neutrinolabs/xrdp For sound using: https://github.com/neutrinolabs/pulseaudio-module-xrdp
Expected behaviour
I expect to hear the call audio over the xrdp-sink. This does work for a ring test in the linphone settings. However, during a call, even though I can see volume levels active in linphone, the ubuntu-desktop settings show no sound level.
Ring - works properly (I can hear it over xrdp, and you can see the output levels represented)
Call - does not work (I cannot hear it, and there is no activity in output levels)
To Reproduce
Build process starting fresh with Ubuntu 22.04 image:
Recreate steps:
Note, similar to the ring test, audio over xrdp works fine for other apps such as firefox or vlc. But, for some reason not for SIP call RTP audio.
Additional context
Thanks for reading. Actually testing the SIP call is a little more involved so not completely spelled out in the recreate. To clarify this call is direct peer-to-peer SIP on a LAN (albeit a virtual network in public cloud infrastructure). I have another client running a SIPp server script that reads a WAV file and returns the audio, and I am calling to it from the linphone. Another option is to deploy a second instance with sound input and output bridged together, then play a video or audio file there after establishing the call from the linphone. I have done this with an Oracle Linux instance which by default bridges the sound input and output, then play a video on firefox. I am running these on "always free" instances on Oracle cloud so this exact scenario can be recreated at no cost.
SDK logs URL
No response