<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">David,<div><span class="Apple-tab-span" style="white-space:pre"> </span>Your client is sending ulaw (the 0 on the m= line is ulaw) and speex on dynamic payload 98 which is fine. FreeSWITCH picks the first codec on your clients list and 200ok's if you wish to force your codec preferences over the clients you need to set codec-negotiation to greedy so that your in control on the server side and the client will not... currently you have the setting of generous I suspect.</div><div><br></div><div>/b</div><div><br><div><div>On Nov 23, 2010, at 10:22 AM, David Wafula wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; font-family: Helvetica; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; font-size: medium; ">Hi all,<div>How do i get this to work:</div><div><br></div><div>From SIP client, this is what am sending:</div><div>================================</div><div><br></div><div><div>Content-Type: application/sdp</div><div>Contact: <<a href="sip:1000@xx.xx.xx.xx.xx:6060;transport=UDP">sip:1000@xx.xx.xx.xx.xx:6060;transport=UDP</a>></div><div><br></div><div>v=0</div><div>o=user1 53655765 2353687637 IN IP4 xx.xx.xx.xx.xx</div><div>s=-</div><div>c=IN IP4 xx.xx.xx.xx</div><div>t=0 0</div><div>m=audio 1935 RTP/AVP 0</div><div>a=rtpmap:98 SPEEX/8000<br><br><br>(and, is 98 even the correct value here?)<br><br></div><div><br>but freeswitc logs show this:<br>============================<br>...<br>2010-11-23 15:54:19.947415 [DEBUG] sofia_glue.c:2741 Set Codec<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1000@xx.xx.xx.xx">sofia/internal/1000@xx.xx.xx.xx</a><span class="Apple-converted-space"> </span>PCMU/8000 20 ms 160 samples 64000 bits<br>.....<br><br>2010-11-23 15:54:19.993146 [DEBUG] mod_sofia.c:683 Local SDP<span class="Apple-converted-space"> </span><a href="mailto:sofia/internal/1000@xx.xx.xx.xx">sofia/internal/1000@xx.xx.xx.xx</a>:<br>v=0<br>o=FreeSWITCH 1290496453 1290496454 IN IP4 xx.xx.xx.xx<br>s=FreeSWITCH<br>c=IN IP4 xx.xx.xx.xx<br>t=0 0<br>m=audio 31206 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br>.....<br><br></div><div>Why is it sticking to PCMU/8000?<br><br>and though RTP flows, audio not working.<br></div><div>in vars.xml, i have:<br><br>...<br> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@8000h@20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/><br> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@8000h@20i,PCMU,PCMA,GSM"/><br>.....<br><br></div></div></span></blockquote></div><br></div></body></html>