<div dir="ltr">Hello,<br><br>SIP first of all uses UDP most of the times (and rarely TCP) .<br>It only create a tunnel that other protocols such as RDP and SDP are going inside.<br><br>Can you explain a bit more what exactly do you wish to implement (i'm not sure that I understand) ?<br>
<br>Ido<br><br clear="all"><a href="http://ik.homelinux.org/">http://ik.homelinux.org/</a><br>
<br><br><div class="gmail_quote">On Tue, Jan 5, 2010 at 15:08, lloyd thomas <span dir="ltr"><<a href="mailto:lloydie.t@googlemail.com">lloydie.t@googlemail.com</a>></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;">
Please forgive my ignorance regarding the project I am about to embark on.<br>Need some advice on Lnet and capturing SIP RTP data to file. In the first instance I just need to work with the sip messages. I believe the SIP messages are similar to HTTP. I am using wireshark to understand how the SIP messages are processed, but I am not sure how to capture and read each frame (wireshark speak) using Lnet.<br>
<br>(FYI: <a href="http://en.wikipedia.org/wiki/Session_Initiation_Protocol#SIP_network_elements" target="_blank">http://en.wikipedia.org/wiki/Session_Initiation_Protocol#SIP_network_elements</a>)<br><br>For instance the following is from frame 12 (935 bytes)<br>
----------------------------------------------------------------------------------------------<br>INVITE <a href="mailto:sip%3A1002@192.168.91.200" target="_blank">sip:1002@192.168.91.200</a> SIP/2.0<br>Via:SIP/2.0/UDP 192.168.91.190:5060;rport;branch=z9hG4bK83570061<br>
From:"Lloyd" <<a href="mailto:sip%3A1000@192.168.91.200" target="_blank">sip:1000@192.168.91.200</a>>;tag=4b428357-2fa-1ec5d4e<br>To:<<a href="mailto:sip%3A1002@192.168.91.200" target="_blank">sip:1002@192.168.91.200</a>><br>
Contact:"Lloyd" <sip:1000@192.168.91.190:5060;transport=UDP><br>
<a href="mailto:Call-ID%3A83570000-4ce59f27@192.168.91.200" target="_blank">Call-ID:83570000-4ce59f27@192.168.91.200</a><br>Subject:sip phone call<br>CSeq:2112045024 INVITE<br>User-Agent:Mitel-5212-SIP-Phone 07.02.00.04 08000F24BEE5<br>
Allow:INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY,PRACK,UPDATE<br>
Allow-Events:talk,hold,conference<br>Supported:timer,100rel,replaces<br>Session-Expires: 1800<br>Min-SE: 90<br>Max-Forwards:70<br>Content-Type:application/sdp<br>Content-Length:247<br><br>v=0<br>o=1000 1262650963 1262650962 IN IP4 192.168.91.190<br>
s=SIP Call<br>c=IN IP4 192.168.91.190<br>t=0 0<br>a=sendrecv<br>m=audio 20036 RTP/AVP 0 8 18 96<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>a=rtpmap:96 telephone-event/8000<br>-----------------------------------------------------------------------------------------------<br>
<br>I get a response from the sip server on frame 13 (371 bytes)<br>----------------------------------------------------------------------------------------------<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.91.190:5060;rport=5060;branch=z9hG4bK83570061<br>
From: "Lloyd" <<a href="mailto:sip%3A1000@192.168.91.200" target="_blank">sip:1000@192.168.91.200</a>>;tag=4b428357-2fa-1ec5d4e<br>To: <<a href="mailto:sip%3A1002@192.168.91.200" target="_blank">sip:1002@192.168.91.200</a>><br>
Call-ID: <a href="mailto:83570000-4ce59f27@192.168.91.200" target="_blank">83570000-4ce59f27@192.168.91.200</a><br>
CSeq: 2112045024 INVITE<br>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15355M<br>Content-Length: 0<br>-----------------------------------------------------------------------------------------------<br><br><br><br>
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