Stefan Sayer writes: > > What kind of tests show bad audio quality if the above change is made?
> try calling into the echo or conference app from e.g. baresip. What > you'll see is that the PLC functions are called all the time because > there's audio 'missing'. I tried echo and didn't notice any problems with voice quality. No error message showed up in syslog or baresip console: call: connecting to 'sip:[email protected]'.. call: SIP Progress: 100 Trying (/) call: SIP Progress: 100 Trying (/) audio: Set audio decoder: opus 48000Hz 2ch audio: Set audio encoder: opus 48000Hz 2ch audio tx pipeline: alsa ---> opus audio rx pipeline: alsa <--- opus [email protected]: Call established: sip:[email protected] Quit0:47] audio=99421/26008 (bit/s) Stopping 1 useragent.. sip:[email protected]: Call with sip:[email protected] terminated (duration: 47 secs) audio Transmit: Receive: packets: 2373 2359 avg. bitrate: 96.0 24.0 (kbit/s) errors: 0 0 I can try conference tomorrow. If channels=1 works for you and 2 does not, then I would suggest that even when channels=1, sems would still be made to generate correct rtpmap line with /2, but without fmtp stereo line. -- Juha _______________________________________________ Semsdev mailing list [email protected] http://lists.iptel.org/mailman/listinfo/semsdev
