haloha
2010-Mar-26 10:10 UTC
[asterisk-users] need help on setup rtp directly between 2 sip clients
Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes when make call between 2 sip clients and see the debug in asterisk console the problem is asterisk setup the inital call for media = asterisk IP address, when all things done, asterisk does re-invite to setup the rtp directly between 2 sip clients is there any way to setup rtp directly between 2 sip clients, no need to go through asterisk server here is my debug log: <--- SIP read from UDP://192.168.1.4:18341 ---> INVITE sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.4:18341 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:test at 192.168.1.4:18341> To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=f543a140Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 261 v=0 o=- 8 2 IN IP4 192.168.1.4 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.4 t=0 0 m=audio 50420 RTP/AVP 107 0 8 101 <--- Transmitting (no NAT) to 192.168.1.4:18341 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.4:18341 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=f543a140To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.26 Supported: replaces, timer Contact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> Content-Length: 0 Reliably Transmitting (no NAT) to 192.168.1.2:34312: INVITE sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport Max-Forwards: 70 From: "Do Nguyen Ha" <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=as2886cf30To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176> Contact: <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>> Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.26 Date: Thu, 25 Mar 2010 12:15:05 GMT Supported: replaces, timer Content-Type: application/sdp Content-Length: 309 v=0 o=root 1983608375 1983608375 IN IP4 192.168.1.5 s=Asterisk PBX 1.6.0.26 c=IN IP4 192.168.1.5 t=0 0 m=audio 17580 RTP/AVP 0 3 8 101 <--- SIP read from UDP://192.168.1.2:34312 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060 Contact: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176> To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=as2886cf30Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <--- Transmitting (no NAT) to 192.168.1.4:18341 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.4:18341 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=f543a140To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.26 Supported: replaces, timer ontact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> Content-Length: 0 <--- SIP read from UDP://192.168.1.2:34312 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060 Contact: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176> To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=as2886cf30Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 CSeq: 102 INVITE Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 183 v=0 o=- 8 2 IN IP4 192.168.1.2 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.2 t=0 0 m=audio 53062 RTP/AVP 0 8 101 <-------------> ACK sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2852e1cc;rport Max-Forwards: 70 From: "Do Nguyen Ha" <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=as2886cf30To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c Contact: <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>> Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.26 Content-Length: 0 <--- Reliably Transmitting (no NAT) to 192.168.1.4:18341 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.4:18341 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=f543a140To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.26 Supported: replaces, timer Contact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> Content-Type: application/sdp Content-Length: 286 v=0 o=root 1290114102 1290114102 IN IP4 192.168.1.5 s=Asterisk PBX 1.6.0.26 c=IN IP4 192.168.1.5 t=0 0 m=audio 18366 RTP/AVP 0 8 101 Reliably Transmitting (no NAT) to 192.168.1.2:34312: INVITE sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport Max-Forwards: 70 From: "Do Nguyen Ha" <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=as2886cf30To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c Contact: <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>> Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.0.26 Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 286 v=0 o=root 1983608375 1983608376 IN IP4 192.168.1.4 s=Asterisk PBX 1.6.0.26 c=IN IP4 192.168.1.4 t=0 0 m=audio 50420 RTP/AVP 0 8 101 <--- SIP read from UDP://192.168.1.2:34312 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport=5060 To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c From: "Do Nguyen Ha" <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=as2886cf30Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 CSeq: 103 INVITE Content-Length: 0 <--- SIP read from UDP://192.168.1.4:18341 ---> ACK sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.4:18341 ;branch=z9hG4bK-d8754z-e104ab75c9163459-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:test at 192.168.1.4:18341> To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=f543a140Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 ACK User-Agent: X-Lite release 1104o stamp 56125 Authorization: Digest username="test",realm="asterisk",nonce="44b4dd5e",uri="sip:1000 at 192.168.1.5<sip%3A1000 at 192.168.1.5> ",response="540173a06f742b7f11cde8010f90ec26",algorithm=MD5 Content-Length: 0 <-------------> Reliably Transmitting (no NAT) to 192.168.1.4:18341: INVITE sip:test at 192.168.1.4:18341 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK584d3aaf;rport Max-Forwards: 70 From: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 To: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=f543a140Contact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.26 Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 286 v=0 o=root 1290114102 1290114103 IN IP4 192.168.1.2 s=Asterisk PBX 1.6.0.26 c=IN IP4 192.168.1.2 t=0 0 m=audio 53062 RTP/AVP 0 8 101 <--- SIP read from UDP://192.168.1.4:18341 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK584d3aaf;rport=5060 Contact: <sip:test at 192.168.1.4:18341> To: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=f543a140From: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 102 INVITE Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 183 v=0 o=- 8 3 IN IP4 192.168.1.4 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.4 t=0 0 m=audio 50420 RTP/AVP 0 8 101 <-------------> Transmitting (no NAT) to 192.168.1.4:18341: ACK sip:test at 192.168.1.4:18341 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK5dde1d6e;rport Max-Forwards: 70 From: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 To: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=f543a140Contact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.26 Content-Length: 0 <--- SIP read from UDP://192.168.1.2:34312 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport=5060 Contact: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176> To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>>;tag=as2886cf30Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 CSeq: 103 INVITE Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 183 v=0 o=- 8 2 IN IP4 192.168.1.2 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.2 t=0 0 m=audio 53062 RTP/AVP 0 8 101 Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100326/7635d515/attachment.htm
Alyed
2010-Mar-26 16:56 UTC
[asterisk-users] need help on setup rtp directly between 2 sip clients
I guess to do what you want you need to dial directly between the phones. Can't do it with xlite but you can with SJphones Don't remember the exact syntax but guess it's something like sip:username at the.phones.ip:5060 Alyed 2010/3/26 haloha <haloha201 at gmail.com>> Hi all > > my asterisk server, 2 sip client softphones are the same LAN > > asterisk ip address : 192.168.1.5 > sip client 1 : 192.168.1.4 > sip client 2 : 192.168.1.2 > > asterisk starts ok with sip > > setup the sip.conf > [test] > type=friend > username=test > secret=1000 > host=dynamic > context=cucku > directmedia=yes > directrtpsetup=yes > > [1000] > type=friend > username=1000 > secret=1000 > host=dynamic > context=cucku > directmedia=yes > directrtpsetup=yes > > when make call between 2 sip clients and see the debug in asterisk console > the problem is asterisk setup the inital call for media = asterisk IP > address, when all things done, asterisk does re-invite to setup the rtp > directly between 2 sip clients > > is there any way to setup rtp directly between 2 sip clients, no need to go > through asterisk server > > here is my debug log: > <--- SIP read from UDP://192.168.1.4:18341 ---> > INVITE sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5> SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.4:18341 > ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;rport > Max-Forwards: 70 > Contact: <sip:test at 192.168.1.4:18341> > To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> > From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=f543a140 > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 2 INVITE > Content-Type: application/sdp > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 261 > v=0 > o=- 8 2 IN IP4 192.168.1.4 > s=CounterPath X-Lite 3.0 > c=IN IP4 192.168.1.4 > t=0 0 > m=audio 50420 RTP/AVP 107 0 8 101 > > <--- Transmitting (no NAT) to 192.168.1.4:18341 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.4:18341 > ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 > From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=f543a140 > To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 2 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Supported: replaces, timer > Contact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> > Content-Length: 0 > > Reliably Transmitting (no NAT) to 192.168.1.2:34312: > INVITE sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport > Max-Forwards: 70 > From: "Do Nguyen Ha" <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=as2886cf30 > To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176> > Contact: <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>> > Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 > CSeq: 102 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Date: Thu, 25 Mar 2010 12:15:05 GMT > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 309 > v=0 > o=root 1983608375 1983608375 IN IP4 192.168.1.5 > s=Asterisk PBX 1.6.0.26 > c=IN IP4 192.168.1.5 > t=0 0 > m=audio 17580 RTP/AVP 0 3 8 101 > > <--- SIP read from UDP://192.168.1.2:34312 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060 > Contact: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176> > To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c > From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=as2886cf30 > Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 > CSeq: 102 INVITE > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 0 > > > <--- Transmitting (no NAT) to 192.168.1.4:18341 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.4:18341 > ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 > From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=f543a140 > To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 2 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Supported: replaces, timer > ontact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> > Content-Length: 0 > > > <--- SIP read from UDP://192.168.1.2:34312 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2dfa6cce;rport=5060 > Contact: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176> > To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c > From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=as2886cf30 > Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 > CSeq: 102 INVITE > Content-Type: application/sdp > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 183 > v=0 > o=- 8 2 IN IP4 192.168.1.2 > s=CounterPath X-Lite 3.0 > c=IN IP4 192.168.1.2 > t=0 0 > m=audio 53062 RTP/AVP 0 8 101 > > <-------------> > ACK sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK2852e1cc;rport > Max-Forwards: 70 > From: "Do Nguyen Ha" <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=as2886cf30 > To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c > Contact: <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>> > Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 > CSeq: 102 ACK > User-Agent: Asterisk PBX 1.6.0.26 > Content-Length: 0 > > <--- Reliably Transmitting (no NAT) to 192.168.1.4:18341 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.4:18341 > ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 > From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=f543a140 > To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 2 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Supported: replaces, timer > Contact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> > Content-Type: application/sdp > Content-Length: 286 > v=0 > o=root 1290114102 1290114102 IN IP4 192.168.1.5 > s=Asterisk PBX 1.6.0.26 > c=IN IP4 192.168.1.5 > t=0 0 > m=audio 18366 RTP/AVP 0 8 101 > > > Reliably Transmitting (no NAT) to 192.168.1.2:34312: > INVITE sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport > Max-Forwards: 70 > From: "Do Nguyen Ha" <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=as2886cf30 > To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c > Contact: <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5>> > Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 > CSeq: 103 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Supported: replaces, timer > X-asterisk-Info: SIP re-invite (External RTP bridge) > Content-Type: application/sdp > Content-Length: 286 > v=0 > o=root 1983608375 1983608376 IN IP4 192.168.1.4 > s=Asterisk PBX 1.6.0.26 > c=IN IP4 192.168.1.4 > t=0 0 > m=audio 50420 RTP/AVP 0 8 101 > > <--- SIP read from UDP://192.168.1.2:34312 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport=5060 > To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c > From: "Do Nguyen Ha" <sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=as2886cf30 > Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 > CSeq: 103 INVITE > Content-Length: 0 > > > <--- SIP read from UDP://192.168.1.4:18341 ---> > ACK sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5> SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.4:18341 > ;branch=z9hG4bK-d8754z-e104ab75c9163459-1---d8754z-;rport > Max-Forwards: 70 > Contact: <sip:test at 192.168.1.4:18341> > To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 > From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=f543a140 > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 2 ACK > User-Agent: X-Lite release 1104o stamp 56125 > Authorization: Digest > username="test",realm="asterisk",nonce="44b4dd5e",uri=" > sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5> > ",response="540173a06f742b7f11cde8010f90ec26",algorithm=MD5 > Content-Length: 0 > > > <-------------> > Reliably Transmitting (no NAT) to 192.168.1.4:18341: > INVITE sip:test at 192.168.1.4:18341 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK584d3aaf;rport > Max-Forwards: 70 > From: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 > To: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=f543a140 > Contact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 102 INVITE > User-Agent: Asterisk PBX 1.6.0.26 > Supported: replaces, timer > X-asterisk-Info: SIP re-invite (External RTP bridge) > Content-Type: application/sdp > Content-Length: 286 > v=0 > o=root 1290114102 1290114103 IN IP4 192.168.1.2 > s=Asterisk PBX 1.6.0.26 > c=IN IP4 192.168.1.2 > t=0 0 > m=audio 53062 RTP/AVP 0 8 101 > > <--- SIP read from UDP://192.168.1.4:18341 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK584d3aaf;rport=5060 > Contact: <sip:test at 192.168.1.4:18341> > To: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=f543a140 > From: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 102 INVITE > Content-Type: application/sdp > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 183 > v=0 > o=- 8 3 IN IP4 192.168.1.4 > s=CounterPath X-Lite 3.0 > c=IN IP4 192.168.1.4 > t=0 0 > m=audio 50420 RTP/AVP 0 8 101 > > <-------------> > Transmitting (no NAT) to 192.168.1.4:18341: > ACK sip:test at 192.168.1.4:18341 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK5dde1d6e;rport > Max-Forwards: 70 > From: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>>;tag=as0307d0b3 > To: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=f543a140 > Contact: <sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> > Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. > CSeq: 102 ACK > User-Agent: Asterisk PBX 1.6.0.26 > Content-Length: 0 > > <--- SIP read from UDP://192.168.1.2:34312 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK39e834a9;rport=5060 > Contact: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176> > To: <sip:1000 at 192.168.1.2:34312;rinstance=862211afcf483176>;tag=403c255c > From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> > >;tag=as2886cf30 > Call-ID: 5fcaa0dc5dd8cac86f01164b2ea6d03a at 192.168.1.5 > CSeq: 103 INVITE > Content-Type: application/sdp > User-Agent: X-Lite release 1104o stamp 56125 > Content-Length: 183 > v=0 > o=- 8 2 IN IP4 192.168.1.2 > s=CounterPath X-Lite 3.0 > c=IN IP4 192.168.1.2 > t=0 0 > m=audio 53062 RTP/AVP 0 8 101 > > Thank you > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100326/0eb1bccd/attachment.htm
Juan E. RodrÃguez
2010-Mar-27 06:37 UTC
[asterisk-users] need help on setup rtp directly between 2 sipclients
Try setting canreinvite and nat to no for those extensions.
Saludos,
Juan E. Rodr?guez
-----Original Message-----
From: Alyed <alyed at vivoxie.com>
Date: Fri, 26 Mar 2010 10:56:50
To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at
lists.digium.com>
Subject: Re: [asterisk-users] need help on setup rtp directly between 2 sip
clients
--
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