I think I found my missing ACKs! Can anyone tell me why they work be being sent to the loopback interface? The destination address is still the external (eth0) IP.
If it is eth0 of the same server, then is the kernel sending via loopback interface at it detects the destination is itself.
You should paste here ngrep with the sip traffic from invite to bye in order to give more hints about what is going wrong there.
Cheers, Daniel
On 8/28/13 3:09 PM, Marc Soda wrote:
I think I found my missing ACKs! Can anyone tell me why they work be being sent to the loopback interface? The destination address is still the external (eth0) IP.
--
Marc Soda, Sr. Systems Engineer *CoreDial, LLC* | www.coredial.com http://www.coredial.com
1787 Sentry Parkway West, Building 16, Suite 100, Blue Bell, PA 19422 Office: (215) 297-4400 x203 | Fax: (215) 297-4401 | Email: msoda@coredial.com mailto:msoda@coredial.com
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Thanks, I appreciate it.
In this setup the there are 2 endpoints (700 and 701) peered up to an Asterisk server (172.16.60.6) via a Kamailio proxy (172.16.60.20). 700 (172.16.60.28) is calling 701 (172.16.3.65). When 701 answers the OK is sent to the proxy and then to Asterisk. Asterisk is then ACKing the OK. The ACK is being sent to the proxy and then the proxy should be sending it back to the endpoint. It is not. The ACK is being sent to the proxy and then the proxy is sending to itself again, via the loopback interface. I believe loose_route() should be re-writing the destination to be the endpoint, but it not.
Trace:
U 172.16.60.28:54936 -> 172.16.60.20:5060 INVITE sip:701@eng-reg1.example.com SIP/2.0. Via: SIP/2.0/UDP 172.16.60.28:54936 ;rport;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44. Max-Forwards: 70. From: <sip:sip700_tbs@eng-reg1.example.com
;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: sip:701@eng-reg1.example.com. Contact: sip:sip700_tbs@172.16.60.28:54936;ob. Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi. CSeq: 13862 INVITE. Route: sip:eng-reg1.example.com;transport=udp;lr. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_mecha-15/r2272. Proxy-Authorization: Digest username="sip700_tbs", realm="sip700_tbs", nonce="Uh39eVId/E2Vxz5hgWC/7jMNGAf7rxrV", uri="sip:701@eng-reg1.example.com", response="8a74a8727baa45df84ea1374cb6668f2", cnonce="EYGYpa1zWmEXcOMighUzGZ20cY2HJ7AJ", qop=auth, nc=00000001. Content-Type: application/sdp. Content-Length: 340. . v=0. o=- 3586685645 3586685645 IN IP4 172.16.60.28. s=pjmedia. c=IN IP4 172.16.60.28. t=0 0. m=audio 4004 RTP/AVP 99 0 8 101. c=IN IP4 172.16.60.28. a=rtcp:4005 IN IP4 172.16.60.28. a=sendrecv. a=rtpmap:99 SILK/24000. a=fmtp:99 useinbandfec=0. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
U 172.16.60.20:5060 -> 172.16.60.28:54936 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.60.28:54936 ;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44. From: <sip:sip700_tbs@eng-reg1.example.com
;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: sip:701@eng-reg1.example.com. Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi. CSeq: 13862 INVITE. Server: kamailio (4.0.3 (x86_64/linux)). Content-Length: 0. .
U 172.16.60.20:5060 -> 172.16.60.6:5060 INVITE sip:701@eng-reg1.example.com SIP/2.0. Record-Route: sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes. Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKe51f.d2338795.0. Via: SIP/2.0/UDP 172.16.60.28:54936 ;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44. Max-Forwards: 16. From: <sip:sip700_tbs@eng-reg1.example.com
;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: sip:701@eng-reg1.example.com. Contact: sip:sip700_tbs@172.16.60.28:54936;ob. Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi. CSeq: 13862 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Supported: replaces, 100rel, timer, norefersub. Session-Expires: 1800. Min-SE: 90. User-Agent: CSipSimple_mecha-15/r2272. Content-Type: application/sdp. Content-Length: 340. . v=0. o=- 3586685645 3586685645 IN IP4 172.16.60.28. s=pjmedia. c=IN IP4 172.16.60.28. t=0 0. m=audio 4004 RTP/AVP 99 0 8 101. c=IN IP4 172.16.60.28. a=rtcp:4005 IN IP4 172.16.60.28. a=sendrecv. a=rtpmap:99 SILK/24000. a=fmtp:99 useinbandfec=0. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
U 172.16.60.6:5060 -> 172.16.60.20:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKe51f.d2338795.0;received=172.16.60.20;rport=5060. Via: SIP/2.0/UDP 172.16.60.28:54936 ;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44. Record-Route: sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes. From: <sip:sip700_tbs@eng-reg1.example.com
;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: sip:701@eng-reg1.example.com. Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi. CSeq: 13862 INVITE. Server: Asterisk1.8. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:701@172.16.60.6:5060. Content-Length: 0. .
U 172.16.60.6:5060 -> 172.16.60.20:5060 INVITE sip:sip701_tbs@172.16.60.20:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK0919ead7;rport. Max-Forwards: 70. From: "Alpha" sip:700@172.16.60.6;tag=as5e1a80d8. To: sip:sip701_tbs@172.16.60.20:5060. Contact: sip:700@172.16.60.6:5060. Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060. CSeq: 102 INVITE. User-Agent: Asterisk1.8. Date: Wed, 28 Aug 2013 13:34:03 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 263. . v=0. o=root 1276916964 1276916964 IN IP4 172.16.60.6. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 172.16.60.6. t=0 0. m=audio 21930 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
U 172.16.60.20:5060 -> 172.16.60.6:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK0919ead7;rport=5060. From: "Alpha" sip:700@172.16.60.6;tag=as5e1a80d8. To: sip:sip701_tbs@172.16.60.20:5060. Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060. CSeq: 102 INVITE. Server: kamailio (4.0.3 (x86_64/linux)). Content-Length: 0. .
U 172.16.60.20:5060 -> 172.16.3.65:5060 INVITE sip:sip701_tbs@172.16.3.65:5060 SIP/2.0. Record-Route: sip:172.16.60.20;lr=on;ftag=as5e1a80d8;nat=yes. Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bK9381.d7d662b.0. Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK0919ead7;rport=5060. Max-Forwards: 16. From: "Alpha" sip:700@172.16.60.6;tag=as5e1a80d8. To: sip:sip701_tbs@172.16.60.20:5060. Contact: sip:700@172.16.60.6:5060. Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060. CSeq: 102 INVITE. User-Agent: Asterisk1.8. Date: Wed, 28 Aug 2013 13:34:03 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 263. . v=0. o=root 1276916964 1276916964 IN IP4 172.16.60.6. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 172.16.60.6. t=0 0. m=audio 21930 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
U 172.16.3.65:5060 -> 172.16.60.20:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.16.60.20;received=172.16.60.20;branch=z9hG4bK9381.d7d662b.0. Via: SIP/2.0/UDP 172.16.60.6:5060;rport=5060;branch=z9hG4bK0919ead7. Record-Route: sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes. Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060. From: "Alpha" sip:700@172.16.60.6;tag=as5e1a80d8. To: sip:sip701_tbs@172.16.60.20;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc. CSeq: 102 INVITE. Contact: sip:sip701_tbs@172.16.60.20:5060. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. Content-Length: 0. .
U 172.16.60.20:5060 -> 172.16.60.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.16.60.6:5060;rport=5060;branch=z9hG4bK0919ead7. Record-Route: sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes. Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060. From: "Alpha" sip:700@172.16.60.6;tag=as5e1a80d8. To: sip:sip701_tbs@172.16.60.20;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc. CSeq: 102 INVITE. Contact: sip:sip701_tbs@172.16.60.20:5060;alias=172.16.3.65~5060~1. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. Content-Length: 0. .
U 172.16.60.6:5060 -> 172.16.60.20:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKe51f.d2338795.0;received=172.16.60.20;rport=5060. Via: SIP/2.0/UDP 172.16.60.28:54936 ;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44. Record-Route: sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes. From: <sip:sip700_tbs@eng-reg1.example.com
;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: sip:701@eng-reg1.example.com;tag=as7e09c7c3. Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi. CSeq: 13862 INVITE. Server: Asterisk1.8. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:701@172.16.60.6:5060. Content-Length: 0. .
U 172.16.60.20:5060 -> 172.16.60.28:54936 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.16.60.28:54936 ;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44. Record-Route: sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes. From: <sip:sip700_tbs@eng-reg1.example.com
;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: sip:701@eng-reg1.example.com;tag=as7e09c7c3. Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi. CSeq: 13862 INVITE. Server: Asterisk1.8. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:701@172.16.60.6:5060. Content-Length: 0. .
U 172.16.3.65:5060 -> 172.16.60.20:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.60.20;received=172.16.60.20;branch=z9hG4bK9381.d7d662b.0. Via: SIP/2.0/UDP 172.16.60.6:5060;rport=5060;branch=z9hG4bK0919ead7. Record-Route: sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes. Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060. From: "Alpha" sip:700@172.16.60.6;tag=as5e1a80d8. To: sip:sip701_tbs@172.16.60.20;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc. CSeq: 102 INVITE. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. Contact: sip:sip701_tbs@172.16.60.20:5060. Supported: replaces, 100rel. Content-Type: application/sdp. Content-Length: 200. . v=0. o=dresden 3586685646 1 IN IP4 172.16.3.65. s=sflphone. c=IN IP4 172.16.3.65. t=0 0. m=audio 41394 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=sendrecv. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
U 172.16.60.20:5060 -> 172.16.60.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.60.6:5060;rport=5060;branch=z9hG4bK0919ead7. Record-Route: sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes. Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060. From: "Alpha" sip:700@172.16.60.6;tag=as5e1a80d8. To: sip:sip701_tbs@172.16.60.20;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc. CSeq: 102 INVITE. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. Contact: sip:sip701_tbs@172.16.60.20:5060;alias=172.16.3.65~5060~1. Supported: replaces, 100rel. Content-Type: application/sdp. Content-Length: 200. . v=0. o=dresden 3586685646 1 IN IP4 172.16.3.65. s=sflphone. c=IN IP4 172.16.3.65. t=0 0. m=audio 41394 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=sendrecv. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
===== This ACK is never being sent back to the endpoint that answered the call (172.16.3.65): =====
U 172.16.60.6:5060 -> 172.16.60.20:5060 ACK sip:sip701_tbs@172.16.60.20:5060;alias=172.16.3.65~5060~1 SIP/2.0. Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK70890881;rport. Route: sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes. Max-Forwards: 70. From: "Alpha" sip:700@172.16.60.6;tag=as5e1a80d8. To: <sip:sip701_tbs@172.16.60.20:5060
;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc.
Contact: sip:700@172.16.60.6:5060. Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060. CSeq: 102 ACK. User-Agent: Asterisk1.8. Content-Length: 0. .
U 172.16.60.6:5060 -> 172.16.60.20:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKe51f.d2338795.0;received=172.16.60.20;rport=5060. Via: SIP/2.0/UDP 172.16.60.28:54936 ;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44. Record-Route: sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes. From: <sip:sip700_tbs@eng-reg1.example.com
;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: sip:701@eng-reg1.example.com;tag=as7e09c7c3. Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi. CSeq: 13862 INVITE. Server: Asterisk1.8. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:701@172.16.60.6:5060. Content-Type: application/sdp. Content-Length: 261. . v=0. o=root 115762304 115762304 IN IP4 172.16.60.6. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 172.16.60.6. t=0 0. m=audio 24198 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
U 172.16.60.20:5060 -> 172.16.60.28:54936 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.60.28:54936 ;rport=54936;branch=z9hG4bKPjBsmwQX-oWu6srxf09JRx98W5IklAwE44. Record-Route: sip:172.16.60.20;lr=on;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes. From: <sip:sip700_tbs@eng-reg1.example.com
;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: sip:701@eng-reg1.example.com;tag=as7e09c7c3. Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi. CSeq: 13862 INVITE. Server: Asterisk1.8. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:701@172.16.60.6:5060. Content-Type: application/sdp. Content-Length: 261. . v=0. o=root 115762304 115762304 IN IP4 172.16.60.6. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 172.16.60.6. t=0 0. m=audio 24198 RTP/AVP 0 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
===== This is the ACK that shows up on the loopback interface (I believe it should be going to 172.16.3.65): =====
U 172.16.60.20:5060 -> 172.16.60.20:5060 ACK sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes SIP/2.0. Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 172.16.60.6:5060;branch=z9hG4bK70890881;rport=5060. Max-Forwards: 16. From: "Alpha" sip:700@172.16.60.6;tag=as5e1a80d8. To: <sip:sip701_tbs@172.16.60.20:5060
;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc.
Contact: sip:700@172.16.60.6:5060. Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060. CSeq: 102 ACK. User-Agent: Asterisk1.8. Content-Length: 0. .
U 172.16.60.28:54936 -> 172.16.60.20:5060 ACK sip:701@172.16.60.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.60.28:54936 ;rport;branch=z9hG4bKPjK4X6qtfWQUXsbgPP8D6j-bxR0KBd7r-x. Max-Forwards: 70. From: <sip:sip700_tbs@eng-reg1.example.com
;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: sip:701@eng-reg1.example.com;tag=as7e09c7c3. Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi. CSeq: 13862 ACK. Route: sip:172.16.60.20;lr;ftag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P;nat=yes. Content-Length: 0. .
U 172.16.60.20:5060 -> 172.16.60.6:5060 ACK sip:701@172.16.60.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.60.20;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 172.16.60.28:54936 ;rport=54936;branch=z9hG4bKPjK4X6qtfWQUXsbgPP8D6j-bxR0KBd7r-x. Max-Forwards: 16. From: <sip:sip700_tbs@eng-reg1.example.com
;tag=bmiDmchqpEnJnCFuHdLZNICQ-DisG41P.
To: sip:701@eng-reg1.example.com;tag=as7e09c7c3. Call-ID: BUDvyYQczBQPRIOhj7t7FodP7675QPGi. CSeq: 13862 ACK. Content-Length: 0. .
Hello,
On 8/28/13 8:22 PM, Marc Soda wrote:
Thanks, I appreciate it.
In this setup the there are 2 endpoints (700 and 701) peered up to an Asterisk server (172.16.60.6) via a Kamailio proxy (172.16.60.20). 700 (172.16.60.28) is calling 701 (172.16.3.65). When 701 answers the OK is sent to the proxy and then to Asterisk. Asterisk is then ACKing the OK. The ACK is being sent to the proxy and then the proxy should be sending it back to the endpoint. It is not. The ACK is being sent to the proxy and then the proxy is sending to itself again, via the loopback interface. I believe loose_route() should be re-writing the destination to be the endpoint, but it not.
what device is at 701? The 200ok receved from it has the contact address with the IP of kamailio:
U 172.16.3.65:5060 -> 172.16.60.20:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.60.20;received=172.16.60.20;branch=z9hG4bK9381.d7d662b.0. Via: SIP/2.0/UDP 172.16.60.6:5060;rport=5060;branch=z9hG4bK0919ead7. Record-Route: sip:172.16.60.20;lr;ftag=as5e1a80d8;nat=yes. Call-ID: 64a513d30fc6a51e54e8255b7169345c@172.16.60.6:5060. From: "Alpha" sip:700@172.16.60.6;tag=as5e1a80d8. To: sip:sip701_tbs@172.16.60.20;tag=64c37e7e-19b8-46ae-ad0d-f2a4abe667bc. CSeq: 102 INVITE. Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE. *Contact: sip:sip701_tbs@172.16.60.20:5060*. .....
It should be its IP address, otherwise Kamailio doesn't know where to sent requests within dialog.
It seems there is a NAT between your kamailio and 701, as kamailio adds alias parameter to Contact in 200ok. That can be used to route the ack, like:
handle_ruri_alias(); $ru = $du; $du = $null;
Do the above for the ACK before calling loose_route(). Then it will get out of kamailio, but might not solve the overall routing, if the 701 is not at next hop or it doesn't like the r-uri, because it is not what it has set.
The best is to ask for the proper fix on 701 device.
Cheers, Daniel
On Thu, Aug 29, 2013 at 3:53 AM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
what device is at 701? The 200ok receved from it has the contact address with the IP of kamailio:
SFLPhone
It seems there is a NAT between your kamailio and 701, as kamailio adds alias parameter to Contact in 200ok. That can be used to route the ack, like:
handle_ruri_alias(); $ru = $du; $du = $null;
There is no NAT in this scenario, although NAT support is enabled as we will have to deal with it. Should I still use handle_ruri_alias()?
Thanks, Marc
I ended up doing this right before the call to loose_route():
if ((is_method("ACK") || is_method("BYE")) && !isdsturiset()) { handle_ruri_alias(); if ($du != $null) { $ru = $du; $du = $null; } }
Thanks a lot Daniel!
On Thu, Aug 29, 2013 at 10:58 AM, Marc Soda msoda@coredial.com wrote:
On Thu, Aug 29, 2013 at 3:53 AM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
what device is at 701? The 200ok receved from it has the contact address with the IP of kamailio:
SFLPhone
It seems there is a NAT between your kamailio and 701, as kamailio adds alias parameter to Contact in 200ok. That can be used to route the ack, like:
handle_ruri_alias(); $ru = $du; $du = $null;
There is no NAT in this scenario, although NAT support is enabled as we will have to deal with it. Should I still use handle_ruri_alias()?
Thanks, Marc
-
Daniel-Constantin Mierla -
Marc Soda