Hi all, How do i get this to work:
From SIP client, this is what am sending: ================================
Content-Type: application/sdp Contact: <sip:1000@xx.xx.xx.xx.xx:6060;transport=UDP> v=0 o=user1 53655765 2353687637 IN IP4 xx.xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 1935 RTP/AVP 0 a=rtpmap:98 SPEEX/8000 (and, is 98 even the correct value here?) but freeswitc logs show this: ============================ ... 2010-11-23 15:54:19.947415 [DEBUG] sofia_glue.c:2741 Set Codec sofia/internal/1000@xx.xx.xx.xx PCMU/8000 20 ms 160 samples 64000 bits ..... 2010-11-23 15:54:19.993146 [DEBUG] mod_sofia.c:683 Local SDP sofia/internal/1000@xx.xx.xx.xx: v=0 o=FreeSWITCH 1290496453 1290496454 IN IP4 xx.xx.xx.xx s=FreeSWITCH c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 31206 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ..... Why is it sticking to PCMU/8000? and though RTP flows, audio not working. in vars.xml, i have: ... <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@8000h @20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@8000h @20i,PCMU,PCMA,GSM"/> ..... -- David Wafula