Hello everyone, I am trying to get a Java Sip client working with speex/16000. FS sets the codec correctly and then starts sending packets to my client:
2009-04-08 21:46:34 [DEBUG] sofia_glue.c:2732 sofia_glue_negotiate_sdp() Audio Codec Compare [speex:100:16000:0]/[SPEEX:99:16000:20] 2009-04-08 21:46:34 [DEBUG] sofia_glue.c:1857 sofia_glue_tech_set_codec() Set Codec sofia/external5090/[email protected]:5090 SPEEX/16000 20 ms 320 samples When the packets arrive jspeex can't decode them and I started to look at them manually to find out what the problem is. The payload of each RTP packet is 42 bytes and when looking for the "OggS" header I can't find it. Or is the ogg header not needed? Jspeex looks for it and as it can't find it, it stops decoding. Is FS sending one frame per packet? Thank you very much for your help. Best wishes, Phil Ps: Wireshark tells me the following for a sample package: Real-Time Transport Protocol Setup Method: SDP 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 .... = Extension: False .... 0000 = Contributing source identifiers count: 0 0... .... = Marker: False Payload type: speex (100) Sequence number: 9365 Extended sequence number: 74901 Timestamp: 23680 Synchronization Source identifier: 0x004a235c (4858716) Payload: 2C6679456EE347FFD0A3D55B133771A9A100DB639F5B8ED2... Payload is 42 bytes. -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01 _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
