[Freeswitch-users] using speex

[David Wafula]

Hi all,
How do i get this to work:

>From SIP client, this is what am sending:
================================

Content-Type: application/sdp
Contact: <sip:1000 at 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 at 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 at 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 at 8000h
@20i,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex at 8000h
@20i,PCMU,PCMA,GSM"/>
.....


-- 
David Wafula
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