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:[email protected] <sip%[email protected]> 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:[email protected]:18341> To: "1000"<sip:[email protected] <sip%[email protected]>> From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> >;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:[email protected] <sip%[email protected]> >;tag=f543a140 To: "1000"<sip:[email protected] <sip%[email protected]>> Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.26 Supported: replaces, timer Contact: <sip:[email protected] <sip%[email protected]>> Content-Length: 0 Reliably Transmitting (no NAT) to 192.168.1.2:34312: INVITE sip:[email protected]: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:[email protected] <sip%[email protected]> >;tag=as2886cf30 To: <sip:[email protected]:34312;rinstance=862211afcf483176> Contact: <sip:[email protected] <sip%[email protected]>> Call-ID: [email protected] 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:[email protected]:34312;rinstance=862211afcf483176> To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> >;tag=as2886cf30 Call-ID: [email protected] 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:[email protected] <sip%[email protected]> >;tag=f543a140 To: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.26 Supported: replaces, timer ontact: <sip:[email protected] <sip%[email protected]>> 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:[email protected]:34312;rinstance=862211afcf483176> To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> >;tag=as2886cf30 Call-ID: [email protected] 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:[email protected]: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:[email protected] <sip%[email protected]> >;tag=as2886cf30 To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c Contact: <sip:[email protected] <sip%[email protected]>> Call-ID: [email protected] 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:[email protected] <sip%[email protected]> >;tag=f543a140 To: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.26 Supported: replaces, timer Contact: <sip:[email protected] <sip%[email protected]>> 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:[email protected]: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:[email protected] <sip%[email protected]> >;tag=as2886cf30 To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c Contact: <sip:[email protected] <sip%[email protected]>> Call-ID: [email protected] 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:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c From: "Do Nguyen Ha" <sip:[email protected] <sip%[email protected]> >;tag=as2886cf30 Call-ID: [email protected] CSeq: 103 INVITE Content-Length: 0 <--- SIP read from UDP://192.168.1.4:18341 ---> ACK sip:[email protected] <sip%[email protected]> 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:[email protected]:18341> To: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> >;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:[email protected]<sip%[email protected]> ",response="540173a06f742b7f11cde8010f90ec26",algorithm=MD5 Content-Length: 0 <-------------> Reliably Transmitting (no NAT) to 192.168.1.4:18341: INVITE sip:[email protected]:18341 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK584d3aaf;rport Max-Forwards: 70 From: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 To: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> >;tag=f543a140 Contact: <sip:[email protected] <sip%[email protected]>> 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:[email protected]:18341> To: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> >;tag=f543a140 From: "1000"<sip:[email protected] <sip%[email protected]>>;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:[email protected]:18341 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK5dde1d6e;rport Max-Forwards: 70 From: "1000"<sip:[email protected] <sip%[email protected]>>;tag=as0307d0b3 To: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> >;tag=f543a140 Contact: <sip:[email protected] <sip%[email protected]>> 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:[email protected]:34312;rinstance=862211afcf483176> To: <sip:[email protected]:34312;rinstance=862211afcf483176>;tag=403c255c From: "Do Nguyen Ha"<sip:[email protected] <sip%[email protected]> >;tag=as2886cf30 Call-ID: [email protected] 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
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