[SR-Users] RTP Between end points behind NAT or edit headers's sdp?

[Daniel-Constantin Mierla]

Hello,

I don't understand what you want to do, however, if you look to change
content of the sdp, see the nathelper, rtpproxy, mangler, sdpops,
textops, textopsx and siputils modules -- these should give you plenty
of options to play with it to update the sdp.

Cheers,
Daniel

On 20/11/14 18:21, Javier Ricke wrote:
> ------phona A-------kamailio---------asterisk-----
> OPTION 1: configure asterisk or kamailio
> i used asterisk, and install kamailio for traffic RTP can be send
> between end points that behind NAT router, and do not have to go
> through RTP proxy,, plz help!
> i think to the moment install kamailio, headrs'sdp fix IP private, but
> no!, how can fix it plz!!?
> regards!
>
> or 
> OPTION 2: edit sip/sdp
> mi sip/sdp is
> [code]
> <--- SIP read from UDP:152.74.21.12:5060 <http://152.74.21.12:5060> --->
> ACK sip:1001 at 152.74.21.12:6112 <http://sip:1001@152.74.21.12:6112> SIP/2.0
> Via: SIP/2.0/UDP 152.74.21.12;branch=z9hG4bKcydzigwkX
> Via: SIP/2.0/UDP
> 190.164.204.227:41553;branch=z9hG4bK-d8754z-84f73c7e042445de-1---d8754z-;rport=41553
> Max-Forwards: 16
> Contact: <sip:JavierTren at 190.164.204.227:41553;transport=UDP>
> To: <sip:1001 at 152.74.21.12
> <mailto:sip%3A1001 at 152.74.21.12>;transport=UDP>;tag=as6e487bf1
> From: <sip:JavierTren at 152.74.21.12
> <mailto:sip%3AJavierTren at 152.74.21.12>;transport=UDP>;tag=2b8fa52c
> Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY.
> CSeq: 2 ACK
> Proxy-Authorization: Digest
> username="JavierTren",realm="152.74.21.12",nonce="VGFnblRhZkIpJRpScSaEi795VKe4uof0",uri="sip:1001 at 152.74.21.12
> <mailto:sip%3A1001 at 152.74.21.12>;transport=UDP",response="116bf459c22231d0a770534d674b768d",algorithm=MD5
> User-Agent: Zoiper r27147
> Content-Length: 0
>
> <------------->
> --- (12 headers 0 lines) ---
> set_destination: Parsing <sip:152.74.21.12;lr=on;ftag=2b8fa52c> for
> address/port to send to
> set_destination: set destination to 152.74.21.12:5060
> <http://152.74.21.12:5060>
> Audio is at 12064
> Adding codec 100002 (gsm) to SDP
> Adding codec 100003 (ulaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (NAT) to 152.74.21.12:5060
> <http://152.74.21.12:5060>:
> INVITE sip:JavierTren at 190.164.204.227:41553;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 152.74.21.12:6112;branch=z9hG4bK3992226d;rport
> Route: <sip:152.74.21.12;lr=on;ftag=2b8fa52c>
> Max-Forwards: 70
> From: <sip:1001 at 152.74.21.12
> <mailto:sip%3A1001 at 152.74.21.12>;transport=UDP>;tag=as6e487bf1
> To: <sip:JavierTren at 152.74.21.12
> <mailto:sip%3AJavierTren at 152.74.21.12>;transport=UDP>;tag=2b8fa52c
> Contact: <sip:1001 at 152.74.21.12:6112 <http://sip:1001@152.74.21.12:6112>>
> Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY.
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 11.13.0
> Session-Expires: 1800;refresher=uac
> Min-SE: 90
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> X-asterisk-Info: SIP re-invite (External RTP bridge)
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 1842142539 1842142540 IN IP4 192.168.1.8
> s=Asterisk PBX 11.13.0
> c=IN IP4 192.168.1.8
> t=0 0
> m=audio 8000 RTP/AVP 3 0 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> [/code]
>
> and the field o and c i need to IP public y no private,,
> plz any?
>
>
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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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