<div dir="ltr">Anthony,<br><br>Yes I'd figured that out :)<br><br>I couldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday).<br>
<br>I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that.<br><br>I'm hoping the sofia dev can point me in the right direction.<br>
<br>I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that).<br><br>Cheers<br><br>Kirk<br>
<br><br><div class="gmail_quote">2008/8/11 Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span><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">I assume it's "whining" about the SDP parse error encountered.<div class="Ih2E3d"><br>nua(0x8120fe0): INVITE server: error parsing SDP<br><br></div>I do not have the spec memorized but I'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"><div><div></div><div class="Wj3C7c">On Mon, Aug 11, 2008 at 8:30 AM, Kirk Bateman <span dir="ltr"><<a href="mailto:kirk.bateman@gmail.com" target="_blank">kirk.bateman@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="Wj3C7c">
<div dir="ltr">Afternoon everyone,<br><br>I have a bit of a problem with Freeswitch receiving RTP INVITEs from my SIP provider (tesco's internet phone ... I know SIP isn't supported technically, but it sort of works...)<br>
<br>Freeswitch (sofia) seems to be whinging about the INVITE SDP format ... but I'm not sure why ... here's a dump of my log / trace, anyone got any clues, I've tried adding extra codecs to my list and setting the late negotiation but that doesn't seem to fix it ... (registration all works fine by the way ...)<br>
<br> ------------------------------------------------------------------------<br> INVITE sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0<br> Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1<br>
Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022<br> CSeq: 100 INVITE<br> From: "ANOTHER_PHONE_NUMBER" <<a href="mailto:sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone>;tag=SDvke8201-b2b.31deb49<br>
To: "MY_PHONE_NUMBER" <<a href="mailto:sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:MY_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone><br> Max-Forwards: 69<br> Content-Type: application/sdp<br>
Contact: <sip:ANOTHER_PHONE_NUMBER@77.75.1.10:5060;transport=udp><br> Allow: INVITE,CANCEL,ACK,BYE<br> Accept: application/sdp<br> Content-Length: 840<br> <br> v=0<br> o=- 3250588766 0 IN IP4 <a href="http://77.75.1.10" target="_blank">77.75.1.10</a><br>
s=-<br> c=IN IP4 <a href="http://77.75.1.10" target="_blank">77.75.1.10</a><br> t=0 0<br> m=audio 54424 RTP/AVP 18 99 101 102 103 15 104 4 105 106 107 108 125 109 100 8 0<br> a=rtpmap:18 G729/8000<br> a=fmtp:18 annexb=no<br>
a=rtpmap:99 G.729a/8000<br> a=rtpmap:101 G.726-16/8000<br> a=rtpmap:102 G.726-24/8000<br> a=rtpmap:103 G.726-32/8000<br> a=rtpmap:15 G728/8000<br> a=rtpmap:104 G.723.1-H/8000<br> a=rtpmap:4 G723/8000<br> a=rtpmap:105 G.723.1-L/8000<br>
a=rtpmap:106 G.729b/8000<br> a=rtpmap:107 G.723.1a-H/8000<br> a=rtpmap:108 G.723.1a-L/8000<br> a=rtpmap:125 G.nX64/8000<br> a=rtpmap:109 AMR/8000<br> a=rtpmap:100 X-NSE/8000<br> a=fmtp:100 192-194,200-202<br>
a=rtpmap:8 PCMA/8000<br> a=rtpmap:0 PCMU/8000<br> a=maxptime:20<br> a=maxptime:30<br> a=ptime:20<br> a=ptime:30<br> a=X-cap: 1 audio RTP/AVP 100<br> a=X-cap: 2 image udptl t38<br> a=X-sqn:0<br> a=X-cpar: a=rtpmap:100 X-NSE/8000<br>
a=X-cpar: a=fmtp:100 192-194,200-202<br> ------------------------------------------------------------------------<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> ------------------------------------------------------------------------<br> SIP/2.0 400 Bad Session Description<br>
Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1<br> From: "ANOTHER_PHONE_NUMBER" <<a href="mailto:sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone>;tag=SDvke8201-b2b.31deb49<br>
To: "MY_PHONE_NUMBER" <<a href="mailto:sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:MY_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone>;tag=yjay36ZrycNmD<br> Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022<br>
CSeq: 100 INVITE<br> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9235<br> Accept: application/sdp<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO<br>
Supported: 100rel, timer, precondition, path, replaces<br>
Allow-Events: talk<br> Content-Length: 0<br> <br> ------------------------------------------------------------------------<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> ------------------------------------------------------------------------<br> ACK sip:MY_PHONE_NUMBER@MY_IP_ADDRESS:5060;transport=udp SIP/2.0<br>
Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1<br> CSeq: 100 ACK<br> Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022<br> From: "ANOTHER_PHONE_NUMBER" <<a href="mailto:sip%3AANOTHER_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:ANOTHER_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone>;tag=SDvke8201-b2b.31deb49<br>
To: "MY_PHONE_NUMBER" <<a href="mailto:sip%3AMY_PHONE_NUMBER@sip.tescointernetphone.com" target="_blank">sip:MY_PHONE_NUMBER@sip.tescointernetphone.com</a>;user=phone>;tag=yjay36ZrycNmD<br> Max-Forwards: 69<br>
Content-Length: 0<br>
<br> ------------------------------------------------------------------------<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/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
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