------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 ---> ACK 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@190.164.204.227:41553;transport=UDP To: sip:1001@152.74.21.12;transport=UDP;tag=as6e487bf1 From: sip:JavierTren@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@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 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: INVITE sip:JavierTren@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@152.74.21.12;transport=UDP;tag=as6e487bf1 To: sip:JavierTren@152.74.21.12;transport=UDP;tag=2b8fa52c Contact: 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?
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@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@190.164.204.227:41553;transport=UDP To: <sip:1001@152.74.21.12 mailto:sip%3A1001@152.74.21.12;transport=UDP>;tag=as6e487bf1 From: <sip:JavierTren@152.74.21.12 mailto:sip%3AJavierTren@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@152.74.21.12 mailto:sip%3A1001@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@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@152.74.21.12 mailto:sip%3A1001@152.74.21.12;transport=UDP>;tag=as6e487bf1 To: <sip:JavierTren@152.74.21.12 mailto:sip%3AJavierTren@152.74.21.12;transport=UDP>;tag=2b8fa52c Contact: <sip:1001@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?
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Daniel-Constantin Mierla -
Javier Ricke