Hi everyone
So, I am trying to create a sip client for an embedded device that has no audio capabilities.
Therefore I am implementing a mem capture to a buffer for the audio data.
This works so far on a host system but not on the embedded target system.
Now, on the target system I'm stuck at the call disconnecting as soon as I pick up the phone.
Sniffing the traffic and looking at the logs tells me that my programm disconnects after pickup. (Thus no network/pbx/codec problem I guess)
The log points out this issue:
pjsua_aud_channel_update() failed for call_id 0 media 0: Invalid operation (PJ_EINVALIDOP)
I looked into this it and only found some mentions about missing libraries (https://stackoverflow.com/questions/20312927 Essentially tells me that portaudio is missing but I have it installed?)
and the following unanswered post that somewhat describes my issue: https://stackoverflow.com/questions/53075903
Also using endpoint.audDevManager().setNullDev(); or endpoint.audDevManager().setNoDev(); doesn't change the outcome.
General information:
PJ-Sip Version: 2.10
My Code: https://pastebin.com/8VxVRFAL
CMakeLists.txt: https://pastebin.com/pWUcPUur
Log File: https://pastebin.com/tTbkJQCp
Target Linux kernel version: 4.3
Networking:
Local network only, everything on a single switch, no firewall
Embedded Sip Program's Address: 172.28.178.19
PBX Address: 172.28.176.10
Telephone's Address: 172.28.178.2
Note: For building the programm, I compiled PJ-Sip on the target and copied its shared object files over to the host system in order to link them.
Not the best strategy but it appears to work as I couldn't cross compile the library.
I also looked into creating a null sound device on the os but it seems far more complex than what is probably required.
Would appreciate any help I can get to solve this problem.
Thanks and best regards
Kenneth Mathis
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