You mean the SDP payload? You kind of need that.... c= is used for RTP transmission. o= always confuses me so I will just say it's important at well.
You can put a proxy in the middle and do topology hiding I guess however, that is beyond the scope of this list? Kind Regards, Nick. On 6/12/13, Mickael MONSIEUR <[email protected]> wrote: > Good morning, or Good afternoon! It depends :-) > > I have a standard Asterisk configuration: > > SIP friends (phones) <-----> Asterisk <-----> SIP gateway to > PSTN converter > 80.236.215.61 109.69.217.6 internal IP ( > 10.4.0.10/255.255.255.0) > > When analyzing traffic on a SIP friend/phone I see this: > > > INVITE sip:[email protected]:64946;ob SIP/2.0 > Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport > Max-Forwards: 70 > From: <sip:[email protected]>;tag=as15b47581 > To: "test" <sip:[email protected]>;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh > Contact: <sip:[email protected]> > Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM > CSeq: 102 INVITE > User-Agent: Asterisk > Require: timer > Session-Expires: 1800;refresher=uas > Min-SE: 90 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 217 > > v=0 > o=root 664087974 664087976 IN IP4 10.4.0.10 > s=Asterisk > c=IN IP4 10.4.0.10 > t=0 0 > m=audio 8652 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > > My equipement IP 10.4.0.10 is visible to the user, why? > > Thank you, > Mickael > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
