[fpc-pascal] Help Reading SIP Messages.

[lloyd thomas]

OK. I wish to monitor and record calls between a SIP endpoint and SIP
Gateway. At the moment I am doing that by connecting the SIP endpoint and my
machine to an ethernet hub so that I can see all the traffic.
In the first instance I just need to correctly capture, read and interprete
the SIP messages so that I can make an informed decision which RDP packets
to capture.

Then I suppose I will have an even harder task capturing the RDP packets and
joining both legs together (my coding skills is not great)!

Lloydie T

2010/1/5 ik <idokan at gmail.com>

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