Ich habe ein ähnliches Problem. Auch Android, Die Verbindung kommt zustande, aber keinerlei Audio.
Aus dem studio-link Log:
call: no common audio codecs - rejected
call: media-nat `turn’ established
json: {json: “%H”:%H%sjson: }answering call from sip:2268966@studio-link.de with 200
audio: Set audio decoder: opus 48000Hz 2ch
audio: player started with sample format S16LE
audio: Set audio encoder: opus 48000Hz 2ch
audio: source started with sample format S16LE
audio tx pipeline: alsa —> webapp_mono —> webapp_vumeter —> webapp_record —> opus
audio rx pipeline: alsa <— webapp_vumeter <— opus
bbd1cf17@studio-link.de: Call established: sip:2268966@studio-link.de
json: {json: “%H”:%H%sjson: }call: got re-INVITE (SDP Offer)
call: terminate call ‘PIS7Tn6WtC’ with sip:2268966@studio-link.de
sip:bbd1cf17@studio-link.de: Call with sip:2268966@studio-link.de terminated (duration: 40 secs)
Linphone Audio Settings
Update: Wenn ich Studio-Link -> Linphone rufe, steht im Log:
call: connecting to ‘sip:2268966@studio-link.de;transport=tls’…
json: {json: “%H”:%H%sjson: }call: media-nat `turn’ established
call: SIP Progress: 100 trying – your call is important to us (/)
call: SIP Progress: 180 Ringing (/)
audio: Set audio decoder: opus 48000Hz 2ch
audio: player started with sample format S16LE
audio: Set audio encoder: opus 48000Hz 2ch
audio: source started with sample format S16LE
audio tx pipeline: alsa —> webapp_mono —> webapp_vumeter —> webapp_record —> opus
audio rx pipeline: alsa <— webapp_vumeter <— opus
bbd1cf17@studio-link.de: Call established: sip:2268966@studio-link.de;transport=tls
json: {json: “%H”:%H%sjson: }call: terminate call ‘0bd5b7fab012e2ec’ with sip:2268966@studio-link.de;transport=tls
sip:bbd1cf17@studio-link.de: Call with sip:2268966@studio-link.de;transport=tls terminated (duration: 14 secs)
Aber auch kein Ton.