[SR-Users] Modifying sip packet

[Ovidiu Sas]

Take a look at the nathelper module:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/nathelper.html#id2592146


Regards,
Ovidiu Sas

On Tue, Jun 14, 2011 at 6:52 PM, Lucas Alvarez <lucasaa at gmail.com> wrote:
> Hi, I want to modify  the following sip packet, I want to change the SDP IP
> address by $di, which rule could filter this packet? Something like: [ if
> method("OK") and $di is RFC1918 ] ??? And with which function could I change
> the Ip address in the SDP content? In this case I would like to
> change 172.16.99.2 by 190.244.125.41.
> Thanks in advance,
> Lucas
> T 172.16.99.2:5060 -> 190.244.125.41:9055 [AP]
>  SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 172.16.99.2;branch=z9hG4bK7c5.bfc43456.0;i=1;received=172.16.99.2;rport=5060.
> Via: SIP/2.0/TCP
> 192.168.15.231:1952;received=190.244.125.41;branch=z9hG4bK-d8754z-be53247ec649be5b-1---d8754z-;rport=9055.
> Record-Route: <sip:172.16.99.2;r2=on;lr=on;ftag=b11c6a4e>.
> Record-Route: <sip:172.16.99.2;transport=tcp;r2=on;lr=on;ftag=b11c6a4e>.
> From: "Lucas"<sip:1104 at 67.152.18.231>;tag=b11c6a4e.
> To: <sip:1101 at 67.152.18.231>;tag=as5356a251.
> Call-ID: NzcwZWIxNGE1MDg5ZjVhZWRhYWNlM2RiNmMwM2M4MWM..
> CSeq: 2 INVITE.
> Server: Asterisk PBX 1.8.4.1.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH.
> Supported: replaces, timer.
> Contact: <sip:1101 at 172.16.99.2:5080>.
> Remote-Party-ID: "Gus Office One"
> <sip:1101 at 67.152.18.231>;party=called;privacy=off;screen=no.
> Content-Type: application/sdp.
> Content-Length: 305.
> .
> v=0.
> o=root 1124623741 1124623741 IN IP4 172.16.99.2.
> s=Asterisk PBX 1.8.4.1.
> c=IN IP4 172.16.99.2.
> t=0 0.
> m=audio 13640 RTP/AVP 0 8 18 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
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