[asterisk-users] need help on setup rtp directly between 2 sip clients

Alyed alyed at vivoxie.com
Fri Mar 26 11:56:50 CDT 2010 

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
>
>
> --
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