Olivier,
First advice: never send real addresses on the list. Mask them with something like "ua_1", "ua_2", "proxy_wan" and so on.
For the request:
a) Content-Lenght is not correct: there is 278, should be around 264
BTW - did you notice the ptivate IPs in SDP? They will not cause "400" response. But you will not have audio (depending on scenario). Try some NAT traversal methods.
olivier.taylor wrote:
hello, does anybody have an idea on what cause a 400 bad request on this invite? thanks by advance :) Olivier
INVITE sip:3227470340@finalcut.be:5060 SIP/2.0. Via: SIP/2.0/UDP 10.252.1.143:5060;branch=z9hG4bK-797350176. From: "MultiVoIP" sip:1300@82.146.123.83:5060;tag=226023331. To: sip:3227470340@82.146.123.83:5060. Call-ID: 1014-1383682753@82.146.123.83. CSeq: 2 INVITE. Contact: sip:1300@10.252.1.143:5060. Proxy-Authorization: Digest username="1300@finalcut.be", realm="82.146.123.83", nonce="44198a959011090a83e34359f0758c2d2b110220", uri="sip:3227470340@finalcut.be:5060", response="32cb44fbd81ae37f2506878bf06e7b46", algorithm=MD5. max-forwards: 70. user-agent: NGAVFXS/6.08.DI.06.08Sep 28 2005. Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY. Content-Type: application/sdp. Content-Length: 278. .
v=0.4 o=1300 0 0 IN IP4 10.252.1.143.31 s=SIP Call.11 c=IN IP4 10.252.1.143.22 t=0 0.6 m=audio 5006 RTP/AVP 18 96.27 a=ptime:80.11 a=fmtp:18 annexb=yes.21 a=rtpmap:18 G729/8000/1.24 a=fmtp:96 0-16.15 a=rtpmap:96 telephone-event/8000/1.35 m=audio 5006 RTP/AVP 4.23 a=ptime:90.11 a=rtpmap:4 G723/8000/1.23