[OpenSIPS-Users] Reversed behaviour when setting up opensips with rtpproxy
Bogdan-Andrei Iancu
bogdan at opensips.org
Tue Mar 29 15:04:27 CEST 2011
Hi Boris,
are you sure you properly did the relation between the two interface in
RTPproxy and the "i" and "e" flags in nathelper - maybe you simply
swapped the interfaces (as meaning) between the definition in rtpproxy
and usage in nathelper.
Regards,
Bogdan
Boris Ratner wrote:
> Hi all!
>
> Please tell me know if this behaviour is intentional:
> Problem with proxying rtp:
> UAC receives ip in the UAS' subnet while UAS receives the ip of UAC's
> subnet of rtp proxy by default.
>
>
> IP-Phone is on 10.200.10.195.
>
> Network configuration:
>
> ast1.local <-----------> opensips+rtpproxy <---------> ast2.local
> 192.168.56.3 192.168.56.2/192.168.58.2 192.168.58.3
> 10.200.10.something
> for ip phone.
>
> SIP:
> ast1 configured with outboundproxy .56.2
> ast2 configured with outboundproxy .58.2
> no ip routing is done on the ALG
>
> OpenSIPS 1.6.4:
> configured to rtpproxy_offer(); on INVITE
> and to rtpproxy_answer(); on reply to it.
>
> rtpproxy 1.2.1:
> in bridge mode 192.168.56.2/192.168.58.2
>
> SIP works fine:
> <ast1>
> Reliably Transmitting (no NAT) to 192.168.56.2:5060:
> OPTIONS sip:ast2.local SIP/2.0
> Via: SIP/2.0/UDP 192.168.56.3:5060;branch=z9hG4bK2a985f57;rport
> From: "asterisk" <sip:asterisk at 192.168.56.3>;tag=as7d368e85
> To: <sip:ast2.local>
> Contact: <sip:asterisk at 192.168.56.3>
> Call-ID: 1a0cd5082af8bd525e2071c10edf2920 at 192.168.56.3
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 07 Mar 2011 21:54:17 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Content-Length: 0
>
> <ast2>
> <--- Transmitting (no NAT) to 192.168.58.2:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.58.2;branch=z9hG4bK5054.135a9ff1.0;received=192.168.58.2
> Via: SIP/2.0/UDP
> 192.168.56.3:5060;received=192.168.56.3;branch=z9hG4bK2a985f57;rport=5060
> Record-Route: <sip:192.168.58.2;r2=on;lr=on>
> Record-Route: <sip:192.168.56.2;r2=on;lr=on>
> From: "asterisk" <sip:asterisk at 192.168.56.3>;tag=as7d368e85
> To: <sip:ast2.local>;tag=as63b00332
> Call-ID: 1a0cd5082af8bd525e2071c10edf2920 at 192.168.56.3
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:192.168.58.3>
> Accept: application/sdp
> Content-Length: 0
>
> THE CALL: from ast1 to ast2
>
> <ast1>
> <------------>
> -- Executing [565656 at incoming:1] Dial("SIP/bratner-000000a9",
> "SIP/ast2/565656") in new stack
> Audio is at 192.168.56.3 port 18922
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 192.168.56.2:5060:
> INVITE sip:565656 at ast2.local SIP/2.0
> Via: SIP/2.0/UDP 192.168.56.3:5060;branch=z9hG4bK391f49a8;rport
> From: "Extension 1001" <sip:bratner at 192.168.56.3>;tag=as6826385c
> To: <sip:565656 at ast2.local>
> Contact: <sip:bratner at 192.168.56.3>
> Call-ID: 2d167b8f4108d4467ff3ea4e1cff7218 at 192.168.56.3
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 07 Mar 2011 21:58:03 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 287
>
> v=0
> o=root 19589 19589 IN IP4 192.168.56.3
> s=session
> c=IN IP4 192.168.56.3
> t=0 0
> m=audio 18922 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> <RTP PROXY>
>
> DBUG:handle_command: received command "U
> 2d167b8f4108d4467ff3ea4e1cff7218 at 192.168.56.3 192.168.56.3 18922
> as6826385c;1"
> INFO:handle_command: new session
> 2d167b8f4108d4467ff3ea4e1cff7218 at 192.168.56.3, tag as6826385c;1
> requested, type strong
> INFO:handle_command: BRAT: given remote address 192.168.56.3
> INFO:create_twinlistener: BINDING TO 0.0.0.0
> INFO:create_twinlistener: BINDING TO 0.0.0.0
> INFO:handle_command: new session on a port 50026 created, tag as6826385c;1
> INFO:handle_command: pre-filling caller's address with 192.168.56.3:18922
> DBUG:doreply: sending reply "50026"
>
>
>
> <ast2>
> <--- SIP read from 192.168.58.2:5060 --->
> INVITE sip:565656 at ast2.local SIP/2.0
> Record-Route: <sip:192.168.58.2;r2=on;lr=on>
> Record-Route: <sip:192.168.56.2;r2=on;lr=on>
> Via: SIP/2.0/UDP 192.168.58.2;branch=z9hG4bKc05.3917bd94.0
> Via: SIP/2.0/UDP
> 192.168.56.3:5060;received=192.168.56.3;branch=z9hG4bK391f49a8;rport=5060
> From: "Extension 1001" <sip:bratner at 192.168.56.3>;tag=as6826385c
> To: <sip:565656 at ast2.local>
> Contact: <sip:bratner at 192.168.56.3>
> Call-ID: 2d167b8f4108d4467ff3ea4e1cff7218 at 192.168.56.3
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 69
> Date: Mon, 07 Mar 2011 21:58:03 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 305
> P-hint: thehelldoiknow
>
> v=0
> o=root 19589 19589 IN IP4 192.168.56.3
> s=session
> c=IN IP4 192.168.56.2
> t=0 0
> m=audio 50026 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> a=nortpproxy:yes
>
>
> <--- Reliably Transmitting (no NAT) to 192.168.58.2:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.58.2;branch=z9hG4bKc05.3917bd94.0;received=192.168.58.2
> Via: SIP/2.0/UDP
> 192.168.56.3:5060;received=192.168.56.3;branch=z9hG4bK391f49a8;rport=5060
> Record-Route: <sip:192.168.58.2;r2=on;lr=on>
> Record-Route: <sip:192.168.56.2;r2=on;lr=on>
> From: "Extension 1001" <sip:bratner at 192.168.56.3>;tag=as6826385c
> To: <sip:565656 at ast2.local>;tag=as112d0057
> Call-ID: 2d167b8f4108d4467ff3ea4e1cff7218 at 192.168.56.3
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:565656 at 192.168.58.3>
> Content-Type: application/sdp
> Content-Length: 285
>
> v=0
> o=root 4755 4755 IN IP4 192.168.58.3
> s=session
> c=IN IP4 192.168.58.3
> t=0 0
> m=audio 19452 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>
> <RTPPROXY>
>
> DBUG:handle_command: received command "L
> 2d167b8f4108d4467ff3ea4e1cff7218 at 192.168.56.3 192.168.58.3 19452
> as6826385c;1 as112d0057;1"
> INFO:handle_command: BRAT: given internal address 192.168.58.3
> INFO:create_twinlistener: BINDING TO 0.0.0.0
> INFO:create_twinlistener: BINDING TO 0.0.0.0
> INFO:handle_command: lookup on ports 50026/52556, session timer restarted
> INFO:handle_command: pre-filling callee's address with 192.168.58.3:19452
> DBUG:doreply: sending reply "52556"
>
> <back at ast1>
>
> <--- SIP read from 192.168.56.2:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.56.3:5060;received=192.168.56.3;branch=z9hG4bK391f49a8;rport=5060
> Record-Route: <sip:192.168.58.2;r2=on;lr=on>
> Record-Route: <sip:192.168.56.2;r2=on;lr=on>
> From: "Extension 1001" <sip:bratner at 192.168.56.3>;tag=as6826385c
> To: <sip:565656 at ast2.local>;tag=as112d0057
> Call-ID: 2d167b8f4108d4467ff3ea4e1cff7218 at 192.168.56.3
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:565656 at 192.168.58.3>
> Content-Type: application/sdp
> Content-Length: 303unforce_rtp_proxy();
>
> v=0
> o=root 4755 4755 IN IP4 192.168.58.3
> s=session
> c=IN IP4 192.168.58.2
> t=0 0
> m=audio 52556 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> a=nortpproxy:yes
>
>
>
>
> rtp debug on ast1:
>
> Got RTP packet from 10.200.10.195:24466 (type 18, seq 052716, ts
> 1254981560, len 000020)
> Sent RTP packet to 192.168.58.2:52556 (type 00, seq 062080, ts
> 1254981560, len 000160)
> Got RTP packet from 10.200.10.195:24466 (type 18, seq 052717, ts
> 1254981720, len 000020)
> Sent RTP packet to 192.168.58.2:52556 (type 00, seq 062081, ts
> 1254981720, len 000160)
> Got RTP packet from 10.200.10.195:24466 (type 18, seq 052718, ts
> 1254981880, len 000020)
> Sent RTP packet to 192.168.58.2:52556 (type 00, seq 062082, ts
> 1254981880, len 000160)
> Got RTP packet from 10.200.10.195:24466 (type 18, seq 052719, ts
> 1254982040, len 000020)
> Sent RTP packet to 192.168.58.2:52556 (type 00, seq 062083, ts
> 1254982040, len 000160)
>
> there is no route to .58.2 from ast1.local
>
>
> rtp debug on ast2:
>
> <------------->
> --- (11 headers 0 lines) ---
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005309, ts
> 000320, len 000160)
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005310, ts
> 000480, len 000160)
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005311, ts
> 000640, len 000160)
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005312, ts
> 000800, len 000160)
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005313, ts
> 000960, len 000160)
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005314, ts
> 001120, len 000160)
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005315, ts
> 001280, len 000160)
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005316, ts
> 001440, len 000160)
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005317, ts
> 001600, len 000160)
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005318, ts
> 001760, len 000160)
> Sent RTP packet to 192.168.56.2:50026 (type 00, seq 005319, ts
> 001920, len 000160)
>
> there is no route to .56.2 from this host.
>
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>
>
--
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 2nd May 2011
OpenSIPS solutions and "know-how"
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