<div dir="ltr">I assume it&#39;s &quot;whining&quot; about the SDP parse error encountered.<br>nua(0x8120fe0): INVITE server: error parsing SDP<br><br>I do not have the spec memorized but I&#39;m sure if we show it to the sofia dev he can tell us the specific problem with the sdp.<br>
<br><br><br><br><div class="gmail_quote">On Mon, Aug 11, 2008 at 8:30 AM, Kirk Bateman <span dir="ltr">&lt;<a href="mailto:kirk.bateman@gmail.com">kirk.bateman@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div dir="ltr">Afternoon everyone,<br><br>I have a bit of a problem with Freeswitch receiving RTP INVITEs from my SIP provider (tesco&#39;s internet phone ... I know SIP isn&#39;t supported technically, but it sort of works...)<br>

<br>Freeswitch (sofia) seems to be whinging about the INVITE SDP format ... but I&#39;m not sure why ... here&#39;s a dump of my log / trace, anyone got any clues, I&#39;ve tried adding extra codecs to my list and setting the late negotiation but that doesn&#39;t seem to fix it ... (registration all works fine by the way ...)<br>

<br>&nbsp;&nbsp; ------------------------------------------------------------------------<br>&nbsp;&nbsp; INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0<br>&nbsp;&nbsp; Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1<br>

&nbsp;&nbsp; Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022<br>&nbsp;&nbsp; CSeq: 100 INVITE<br>&nbsp;&nbsp; From: &quot;ANOTHER_PHONE_NUMBER&quot; &lt;<a href="mailto:sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone&gt;;tag=SDvke8201-b2b.31deb49<br>

&nbsp;&nbsp; To: &quot;MY_PHONE_NUMBER&quot; &lt;<a href="mailto:sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:MY_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone&gt;<br>&nbsp;&nbsp; Max-Forwards: 69<br>&nbsp;&nbsp; Content-Type: application/sdp<br>

&nbsp;&nbsp; Contact: &lt;sip:ANOTHER_PHONE_NUMBER@77.75.1.10:5060;transport=udp&gt;<br>&nbsp;&nbsp; Allow: INVITE,CANCEL,ACK,BYE<br>&nbsp;&nbsp; Accept: application/sdp<br>&nbsp;&nbsp; Content-Length: 840<br>&nbsp;&nbsp; <br>&nbsp;&nbsp; v=0<br>&nbsp;&nbsp; o=- 3250588766 0 IN IP4 <a href="http://77.75.1.10" target="_blank">77.75.1.10</a><br>

&nbsp;&nbsp; s=-<br>&nbsp;&nbsp; c=IN IP4 <a href="http://77.75.1.10" target="_blank">77.75.1.10</a><br>&nbsp;&nbsp; t=0 0<br>&nbsp;&nbsp; m=audio 54424 RTP/AVP 18 99 101 102 103 15 104 4 105 106 107 108 125 109 100 8 0<br>&nbsp;&nbsp; a=rtpmap:18 G729/8000<br>&nbsp;&nbsp; a=fmtp:18 annexb=no<br>

&nbsp;&nbsp; a=rtpmap:99 G.729a/8000<br>&nbsp;&nbsp; a=rtpmap:101 G.726-16/8000<br>&nbsp;&nbsp; a=rtpmap:102 G.726-24/8000<br>&nbsp;&nbsp; a=rtpmap:103 G.726-32/8000<br>&nbsp;&nbsp; a=rtpmap:15 G728/8000<br>&nbsp;&nbsp; a=rtpmap:104 G.723.1-H/8000<br>&nbsp;&nbsp; a=rtpmap:4 G723/8000<br>&nbsp;&nbsp; a=rtpmap:105 G.723.1-L/8000<br>

&nbsp;&nbsp; a=rtpmap:106 G.729b/8000<br>&nbsp;&nbsp; a=rtpmap:107 G.723.1a-H/8000<br>&nbsp;&nbsp; a=rtpmap:108 G.723.1a-L/8000<br>&nbsp;&nbsp; a=rtpmap:125 G.nX64/8000<br>&nbsp;&nbsp; a=rtpmap:109 AMR/8000<br>&nbsp;&nbsp; a=rtpmap:100 X-NSE/8000<br>&nbsp;&nbsp; a=fmtp:100 192-194,200-202<br>

&nbsp;&nbsp; a=rtpmap:8 PCMA/8000<br>&nbsp;&nbsp; a=rtpmap:0 PCMU/8000<br>&nbsp;&nbsp; a=maxptime:20<br>&nbsp;&nbsp; a=maxptime:30<br>&nbsp;&nbsp; a=ptime:20<br>&nbsp;&nbsp; a=ptime:30<br>&nbsp;&nbsp; a=X-cap: 1 audio RTP/AVP 100<br>&nbsp;&nbsp; a=X-cap: 2 image udptl t38<br>&nbsp;&nbsp; a=X-sqn:0<br>&nbsp;&nbsp; a=X-cpar: a=rtpmap:100 X-NSE/8000<br>

&nbsp;&nbsp; a=X-cpar: a=fmtp:100 192-194,200-202<br>&nbsp;&nbsp; ------------------------------------------------------------------------<br>tport_deliver(0x80fcbf0): msg 0x8129468 (1416 bytes) from udp/<a href="http://77.75.1.10:5060/sip" target="_blank">77.75.1.10:5060/sip</a> next=(nil)<br>

nta: received INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 (CSeq 100)<br>nta: canonizing sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060 with contact<br>nta: INVITE (100) going to a default leg<br>nta: timer set to 200 ms<br>

nua: nua_stack_process_request: entering<br>nua: nh_create: entering<br>nua: nh_create_handle: entering<br>nua: nua_stack_set_params: entering<br>soa_clone(static::0x80ef908, 0x80eccb8, 0x8120fe0) called<br>soa_set_params(static::0x8111d08, ...) called<br>

nta_leg_tcreate(0x81200c0)<br>soa_init_offer_answer(static::0x8111d08) called<br>soa_set_remote_sdp(static::0x8111d08, (nil), 0x810ed20, 840) called<br>nua(0x8120fe0): INVITE server: error parsing SDP<br>nua: nua_invite_server_respond: entering<br>

tport_tsend(0x80fcbf0) tpn = UDP/<a href="http://77.75.1.10:5060" target="_blank">77.75.1.10:5060</a><br>tport_resolve addrinfo = <a href="http://77.75.1.10:5060" target="_blank">77.75.1.10:5060</a><br>tport_by_addrinfo(0x80fcbf0): not found by name UDP/<a href="http://77.75.1.10:5060" target="_blank">77.75.1.10:5060</a><br>

tport_vsend returned 661<br>send 661 bytes to udp/[<a href="http://77.75.1.10" target="_blank">77.75.1.10</a>]:5060 at 13:16:28.358779:<br>&nbsp;&nbsp; ------------------------------------------------------------------------<br>&nbsp;&nbsp; SIP/2.0 400 Bad Session Description<br>

&nbsp;&nbsp; Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1<br>&nbsp;&nbsp; From: &quot;ANOTHER_PHONE_NUMBER&quot; &lt;<a href="mailto:sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone&gt;;tag=SDvke8201-b2b.31deb49<br>

&nbsp;&nbsp; To: &quot;MY_PHONE_NUMBER&quot; &lt;<a href="mailto:sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:MY_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone&gt;;tag=yjay36ZrycNmD<br>&nbsp;&nbsp; Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022<br>

&nbsp;&nbsp; CSeq: 100 INVITE<br>&nbsp;&nbsp; User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9235<br>&nbsp;&nbsp; Accept: application/sdp<br>&nbsp;&nbsp; Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO<br>
&nbsp;&nbsp; Supported: 100rel, timer, precondition, path, replaces<br>
&nbsp;&nbsp; Allow-Events: talk<br>&nbsp;&nbsp; Content-Length: 0<br>&nbsp;&nbsp; <br>&nbsp;&nbsp; ------------------------------------------------------------------------<br>nta: sent 400 Bad Session Description for INVITE (100)<br>nta_leg_destroy(0x81200c0)<br>

soa_destroy(static::0x8111d08) called<br>tport_wakeup_pri(0x80fcbf0): events IN<br>tport_recv_event(0x80fcbf0)<br>tport_recv_iovec(0x80fcbf0) msg 0x8111d08 from (udp/<a href="http://192.168.1.10:5060" target="_blank">192.168.1.10:5060</a>) has 442 bytes, veclen = 1<br>

recv 442 bytes from udp/[<a href="http://77.75.1.10" target="_blank">77.75.1.10</a>]:5060 at 13:16:28.377628:<br>&nbsp;&nbsp; ------------------------------------------------------------------------<br>&nbsp;&nbsp; ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0<br>

&nbsp;&nbsp; Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1<br>&nbsp;&nbsp; CSeq: 100 ACK<br>&nbsp;&nbsp; Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022<br>&nbsp;&nbsp; From: &quot;ANOTHER_PHONE_NUMBER&quot; &lt;<a href="mailto:sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone&gt;;tag=SDvke8201-b2b.31deb49<br>

&nbsp;&nbsp; To: &quot;MY_PHONE_NUMBER&quot; &lt;<a href="mailto:sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:MY_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone&gt;;tag=yjay36ZrycNmD<br>&nbsp;&nbsp; Max-Forwards: 69<br>
&nbsp;&nbsp; Content-Length: 0<br>
&nbsp;&nbsp; <br>&nbsp;&nbsp; ------------------------------------------------------------------------<br>tport_deliver(0x80fcbf0): msg 0x8111d08 (442 bytes) from udp/<a href="http://77.75.1.10:5060/sip" target="_blank">77.75.1.10:5060/sip</a> next=(nil)<br>

nta: received ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0 (CSeq 100)<br>nta: ACK (100) is going to INVITE (100)<br>nta: timer set next to 4820 ms<br>nta: timer I fired, terminate 400 response<br>incoming_reclaim_all((nil), (nil), 0xb174a1fc)<br>

nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free<br>nta: timer not set<br>sofia loglevel 0<br>API CALL [sofia(loglevel 0)] output:<br>Sofia-sip log level set to [0]<br><br><br><br>Any clues would be welcomed ...<br>

<br><br>Cheers<br><br>Kirk Bateman<br></div>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
<br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br>
<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400<br>
</div>