Anthony Minessale <anthony.minessale@...> writes:
You are pushing RTP up against TDM, RTP can start and stop, TDM cannot, When there is no RTP the TDM will carry on sending the IDLE_FLAG byte over the wire.
I made both the RTP stack and the TDM endpoint from scratch so you should assume I may have some insight.
What I am arguing with... is that if I say a word like 'two' and RTP looks a bit like this. RTP: Wall Time Seq.No Time 'Sound' 0.0 1 0 'T' 20ms* 2 160 'W' 40ms* 3 320 'O' ... some skew 100 100*160 'T' +20ms 101 101*160 'W' +20ms 102 102*160 'O' and on the remote it arrives as: 'TW(?)O'..'T'..'O' something is wrong with the playback speed/queue/buffer of the audio stack. All I ask for is (because I didn't see anything in the wiki about that): 1.) How do I check if the jitter buffer is enabled on a call. 2.) How do I check if arriving RTP packets arrived too late for the speed of the FreeTDM voice queue. More details: I started with a setup like this: RTP/AMR -> FreeSWITCH (sangoma transcoder) -> FreeTDM.. For debugging I changed it to: AMR -> FreeSWITCH -> RTP/PCMA -> FreeSWITCH -> FreeTDM. (SIP on 5060) (SIP on 5062) The idea was that the first FreeSWITCH has to deal with the jitter and the second receives a clean linear in wall clock time RTP stream. My sofia sip profile includes these params: <param name="rtp-timer-name" value="soft"/> <param name="manual-rtp-bugs" value="SEND_LINEAR_TIMESTAMPS|NEVER_SEND_MARKER|\n IGNORE_DTMF_DURATION" /> <param name="rtp-rewrite-timestamps" value="true"/> The resulting output of the frontend FreeSWITCH is not linear though (so no idea what rtp-timer-name soft should do): But the resulting stream has packets like this: P.No Wall Time Codec 723 12.761045 G.711 PCMA, Seq=18533, Time=90720 731 12.841107 G.711 PCMA, Seq=18534, Time=90880 so somehow the expected arrival time was 17.78 but it ended up being 12.84 and I think FreeSWITCH/FreeTDM/STFU do not deal well with that. any ideas? holger * add a little jitter