We have Kamailio-cluster via route53(round-robin) some-domain.net
we have two kamailio with public IP's
phone1 is registered on kam1 phone2 is registered on kam2
when we are calling from phone1 to phone2 callflow looks:
phone1 => kam1 => asterisk => kam1 => t_relay(address of second kamailio:5078) => kam2 => phone2
it works perfectly, but in case when we are using polycom as phone2 - we are getting 404 response from polycom...
*Invite from second kamailio * 2020/01/20 10:31:21.799327 10.199.240.19:5078 -> 37.17.41.5:49811 INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:10.199.240.191:5078;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:10.199.240.135:5078;lr Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" sip:1015@10.199.240.179;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: sip:jyu3xsfkrz6c5qn@10.199.240.135 Contact: sip:asterisk@10.199.240.179:7060;alias=10.199.240.179~7060~1 Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 CSeq: 22619 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Penny" sip:1015@10.199.240.179 Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 463
*Response from POLYCOM * 2020/01/20 10:31:22.054766 37.17.41.5:49811 -> 10.199.240.19:5078 SIP/2.0 400 Bad Request Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" sip:1015@10.199.240.179;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: sip:jyu3xsfkrz6c5qn@10.199.240.135;tag=8BC58304-83D9B045 CSeq: 22619 INVITE Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 Record-Route: sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge, sip:10.199.240.19:5078;r2=on;lr;nat=yes;rtp=bridge, sip:10.199.240.235:5078;lr User-Agent: PolycomVVX-VVX_450-UA/5.8.0.12848 Accept-Language: en Content-Length: 0
Any ideas how to fix it?
-- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
Hi Zhan,
At first glance, it does not appear that anything about the second request is grammatically invalid.
I suspect the problem you are encountering is UDP fragmentation, as explained in my blog post here:
http://www.evaristesys.com/blog/sip-udp-fragmentation-and-kamailio-the-sip-h...
The size of the second INVITE you pasted is 1198 bytes. Add 463 bytes of encapsulated SDP body (Content-Length header), and it's 1661 bytes - over the UDP fragmentation threshold of ~1480 based on an MTU of 1500 bytes.
This is due to the additional "contributions" of the second Kamailio - extra Via and Record-Route headers. Removing these extras probably puts the message length at just under the fragmentation threshold.
Because the receiver does not get a fully reassembled UDP datagram, the message arrives partly formed (first UDP fragment is the only one received), the Polycom's SIP stack is confused.
-- Alex
On Fri, Jan 24, 2020 at 10:34:21AM -0700, Zhan Bazarov wrote:
We have Kamailio-cluster via route53(round-robin) some-domain.net
we have two kamailio with public IP's
phone1 is registered on kam1 phone2 is registered on kam2
when we are calling from phone1 to phone2 callflow looks:
phone1 => kam1 => asterisk => kam1 => t_relay(address of second kamailio:5078) => kam2 => phone2
it works perfectly, but in case when we are using polycom as phone2 - we are getting 404 response from polycom...
*Invite from second kamailio
2020/01/20 10:31:21.799327 10.199.240.19:5078 -> 37.17.41.5:49811 INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:10.199.240.191:5078;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:10.199.240.135:5078;lr Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" sip:1015@10.199.240.179;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: sip:jyu3xsfkrz6c5qn@10.199.240.135 Contact: sip:asterisk@10.199.240.179:7060;alias=10.199.240.179~7060~1 Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 CSeq: 22619 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Penny" sip:1015@10.199.240.179 Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 463
*Response from POLYCOM
2020/01/20 10:31:22.054766 37.17.41.5:49811 -> 10.199.240.19:5078 SIP/2.0 400 Bad Request Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" sip:1015@10.199.240.179;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: sip:jyu3xsfkrz6c5qn@10.199.240.135;tag=8BC58304-83D9B045 CSeq: 22619 INVITE Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 Record-Route: sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge, sip:10.199.240.19:5078;r2=on;lr;nat=yes;rtp=bridge, sip:10.199.240.235:5078;lr User-Agent: PolycomVVX-VVX_450-UA/5.8.0.12848 Accept-Language: en Content-Length: 0
Any ideas how to fix it?
-- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Is it possible to get a log, sip trace or even a capture for wireshark out of the phone?
That may reveal more about what had happened.
-------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk
On 1/25/20 8:20 AM, Alex Balashov wrote:
Hi Zhan,
At first glance, it does not appear that anything about the second request is grammatically invalid.
I suspect the problem you are encountering is UDP fragmentation, as explained in my blog post here:
http://www.evaristesys.com/blog/sip-udp-fragmentation-and-kamailio-the-sip-h...
The size of the second INVITE you pasted is 1198 bytes. Add 463 bytes of encapsulated SDP body (Content-Length header), and it's 1661 bytes - over the UDP fragmentation threshold of ~1480 based on an MTU of 1500 bytes.
This is due to the additional "contributions" of the second Kamailio - extra Via and Record-Route headers. Removing these extras probably puts the message length at just under the fragmentation threshold.
Because the receiver does not get a fully reassembled UDP datagram, the message arrives partly formed (first UDP fragment is the only one received), the Polycom's SIP stack is confused.
-- Alex
On Fri, Jan 24, 2020 at 10:34:21AM -0700, Zhan Bazarov wrote:
We have Kamailio-cluster via route53(round-robin) some-domain.net
we have two kamailio with public IP's
phone1 is registered on kam1 phone2 is registered on kam2
when we are calling from phone1 to phone2 callflow looks:
phone1 => kam1 => asterisk => kam1 => t_relay(address of second kamailio:5078) => kam2 => phone2
it works perfectly, but in case when we are using polycom as phone2 - we are getting 404 response from polycom...
*Invite from second kamailio
2020/01/20 10:31:21.799327 10.199.240.19:5078 -> 37.17.41.5:49811 INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:10.199.240.191:5078;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:10.199.240.135:5078;lr Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" sip:1015@10.199.240.179;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: sip:jyu3xsfkrz6c5qn@10.199.240.135 Contact: sip:asterisk@10.199.240.179:7060;alias=10.199.240.179~7060~1 Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 CSeq: 22619 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Penny" sip:1015@10.199.240.179 Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 463
*Response from POLYCOM
2020/01/20 10:31:22.054766 37.17.41.5:49811 -> 10.199.240.19:5078 SIP/2.0 400 Bad Request Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" sip:1015@10.199.240.179;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: sip:jyu3xsfkrz6c5qn@10.199.240.135;tag=8BC58304-83D9B045 CSeq: 22619 INVITE Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 Record-Route: sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge, sip:10.199.240.19:5078;r2=on;lr;nat=yes;rtp=bridge, sip:10.199.240.235:5078;lr User-Agent: PolycomVVX-VVX_450-UA/5.8.0.12848 Accept-Language: en Content-Length: 0
Any ideas how to fix it?
-- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
hello! Thanks for reply!
So, now I have two INVITE requests with interval in few seconds:
INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: sip:34.202.130.141:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:34.202.130.141:5078;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:3.220.42.184:5078;lr Via: SIP/2.0/TCP some.proxy.net:5078;branch=z9hG4bK24ec.940ea1b24f9ae5d68a6f97928661dbb6.0 Via: SIP/2.0/UDP some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK24ec.73c81b0f32164242a9bf0beb0f96f7bb.0 Via: SIP/2.0/UDP 10.199.240.176:7060;received=10.199.240.176;rport=7060;branch=z9hG4bKPj726e453b-eb1b-4e07-b49d-45703bc42e38 From: "Rick" sip:1022@10.199.240.176;tag=1c95c3be-0c2a-44ef-8715-28735586ad95 To: sip:jyu3xsfkrz6c5qn@10.199.240.135 Contact: sip:asterisk@10.199.240.176:7060;alias=10.199.240.176~7060~1 Call-ID: 63cf568c-eef4-4262-8e6b-db3f6caedfbb CSeq: 23870 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Rick" sip:1022@10.199.240.176 Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 463
v=0 o=- 128351188 128351188 IN IP4 34.206.126.53 s=Asterisk c=IN IP4 34.206.126.53 t=0 0 m=audio 30010 RTP/AVP 0 101 a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:30011 a=ptime:20 a=ice-ufrag:Cw3k5LF5 a=ice-pwd:SyzlcKXcD2CnOT4hOw72yyvq4c a=candidate:aUPmOZbVuTuCQ7B9 1 UDP 2130706431 34.206.126.53 30010 typ host a=candidate:aUPmOZbVuTuCQ7B9 2 UDP 2130706430 34.206.126.53 30011 typ host
2020/01/26 19:07:01.211105 37.17.41.5:25171 -> 10.199.240.19:5078 SIP/2.0 400 Bad Request Via: SIP/2.0/TCP some.proxy.net:5078;branch=z9hG4bK24ec.940ea1b24f9ae5d68a6f97928661dbb6.0 Via: SIP/2.0/UDP some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK24ec.73c81b0f32164242a9bf0beb0f96f7bb.0 Via: SIP/2.0/UDP 10.199.240.176:7060;received=10.199.240.176;rport=7060;branch=z9hG4bKPj726e453b-eb1b-4e07-b49d-45703bc42e38 From: "Rick" sip:1022@10.199.240.176;tag=1c95c3be-0c2a-44ef-8715-28735586ad95 To: sip:jyu3xsfkrz6c5qn@10.199.240.135;tag=F6209FFC-FB2B5291 CSeq: 23870 INVITE Call-ID: 63cf568c-eef4-4262-8e6b-db3f6caedfbb Record-Route: sip:34.202.130.141:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge, sip:34.202.130.141:5078;r2=on;lr;nat=yes;rtp=bridge, sip:3.220.42.184:5078;lr User-Agent: PolycomVVX-VVX_450-UA/5.8.0.13851 Accept-Language: en Content-Length: 0
*And few seconds later: *
2020/01/26 19:07:06.187829 10.199.240.19:5078 -> 37.17.41.5:25171 INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: sip:34.202.130.141:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:34.202.130.141:5078;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:3.220.42.184:5078;lr Via: SIP/2.0/TCP some.proxy.net:5078;branch=z9hG4bK32d3.45058b373afc3bd5897903a83bc0de7d.0 Via: SIP/2.0/UDP some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK32d3.257af0eab041dbd2a240416d1c6e454f.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPjc442ef42-b769-4fdc-81cf-33453aeaae67 From: "Rick" sip:1022@10.199.240.179;tag=b44068d2-d1af-4b56-87c0-2ea334784c94 To: sip:jyu3xsfkrz6c5qn@10.199.240.135 Contact: sip:asterisk@10.199.240.179:7060;alias=10.199.240.179~7060~1 Call-ID: f9f4006d-8df9-4501-a77d-d3f7ba45c327 CSeq: 14602 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Rick" sip:1022@10.199.240.179 Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 461
v=0 o=- 95944032 95944032 IN IP4 34.206.126.53 s=Asterisk c=IN IP4 34.206.126.53 t=0 0 m=audio 30082 RTP/AVP 0 101 a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:30083 a=ptime:20 a=ice-ufrag:qgZIRgJo a=ice-pwd:mT7rDx0qrjOkoC4Yzlodqfl7ma a=candidate:aUPmOZbVuTuCQ7B9 1 UDP 2130706431 34.206.126.53 30082 typ host a=candidate:aUPmOZbVuTuCQ7B9 2 UDP 2130706430 34.206.126.53 30083 typ host
but this one was succeed
2020/01/26 19:07:07.349559 37.17.41.5:25171 -> 10.199.240.19:5078 SIP/2.0 180 Ringing Via: SIP/2.0/TCP some.proxy.net:5078;branch=z9hG4bK32d3.45058b373afc3bd5897903a83bc0de7d.0 Via: SIP/2.0/UDP some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK32d3.257af0eab041dbd2a240416d1c6e454f.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPjc442ef42-b769-4fdc-81cf-33453aeaae67 From: "Rick" sip:1022@10.199.240.179;tag=b44068d2-d1af-4b56-87c0-2ea334784c94 To: "1030_DEV" sip:jyu3xsfkrz6c5qn@10.199.240.135;tag=63F0AFCC-6C807D61 CSeq: 14602 INVITE Call-ID: f9f4006d-8df9-4501-a77d-d3f7ba45c327 Contact: sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp Record-Route: sip:34.202.130.141:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge, sip:34.202.130.141:5078;r2=on;lr;nat=yes;rtp=bridge, sip:3.220.42.184:5078;lr User-Agent: PolycomVVX-VVX_450-UA/5.8.0.13851 Allow-Events: conference,talk,hold Accept-Language: en Call-Info: sip:some.proxy.net;appearance-index=1 Content-Length: 0
-- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
It is certainly possible that I am mistaken about the root cause.
— Sent from my iPad
On Jan 27, 2020, at 2:37 AM, Zhan Bazarov chiefkeeft@gmail.com wrote:
hello! Thanks for reply!
So, now I have two INVITE requests with interval in few seconds:
INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: sip:34.202.130.141:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:34.202.130.141:5078;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:3.220.42.184:5078;lr Via: SIP/2.0/TCP some.proxy.net:5078;branch=z9hG4bK24ec.940ea1b24f9ae5d68a6f97928661dbb6.0 Via: SIP/2.0/UDP some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK24ec.73c81b0f32164242a9bf0beb0f96f7bb.0 Via: SIP/2.0/UDP 10.199.240.176:7060;received=10.199.240.176;rport=7060;branch=z9hG4bKPj726e453b-eb1b-4e07-b49d-45703bc42e38 From: "Rick" sip:1022@10.199.240.176;tag=1c95c3be-0c2a-44ef-8715-28735586ad95 To: sip:jyu3xsfkrz6c5qn@10.199.240.135 Contact: sip:asterisk@10.199.240.176:7060;alias=10.199.240.176~7060~1 Call-ID: 63cf568c-eef4-4262-8e6b-db3f6caedfbb CSeq: 23870 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Rick" sip:1022@10.199.240.176 Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 463
v=0 o=- 128351188 128351188 IN IP4 34.206.126.53 s=Asterisk c=IN IP4 34.206.126.53 t=0 0 m=audio 30010 RTP/AVP 0 101 a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:30011 a=ptime:20 a=ice-ufrag:Cw3k5LF5 a=ice-pwd:SyzlcKXcD2CnOT4hOw72yyvq4c a=candidate:aUPmOZbVuTuCQ7B9 1 UDP 2130706431 34.206.126.53 30010 typ host a=candidate:aUPmOZbVuTuCQ7B9 2 UDP 2130706430 34.206.126.53 30011 typ host
2020/01/26 19:07:01.211105 37.17.41.5:25171 -> 10.199.240.19:5078 SIP/2.0 400 Bad Request Via: SIP/2.0/TCP some.proxy.net:5078;branch=z9hG4bK24ec.940ea1b24f9ae5d68a6f97928661dbb6.0 Via: SIP/2.0/UDP some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK24ec.73c81b0f32164242a9bf0beb0f96f7bb.0 Via: SIP/2.0/UDP 10.199.240.176:7060;received=10.199.240.176;rport=7060;branch=z9hG4bKPj726e453b-eb1b-4e07-b49d-45703bc42e38 From: "Rick" sip:1022@10.199.240.176;tag=1c95c3be-0c2a-44ef-8715-28735586ad95 To: sip:jyu3xsfkrz6c5qn@10.199.240.135;tag=F6209FFC-FB2B5291 CSeq: 23870 INVITE Call-ID: 63cf568c-eef4-4262-8e6b-db3f6caedfbb Record-Route: sip:34.202.130.141:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge, sip:34.202.130.141:5078;r2=on;lr;nat=yes;rtp=bridge, sip:3.220.42.184:5078;lr User-Agent: PolycomVVX-VVX_450-UA/5.8.0.13851 Accept-Language: en Content-Length: 0
*And few seconds later: *
2020/01/26 19:07:06.187829 10.199.240.19:5078 -> 37.17.41.5:25171 INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: sip:34.202.130.141:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:34.202.130.141:5078;r2=on;lr;nat=yes;rtp=bridge Record-Route: sip:3.220.42.184:5078;lr Via: SIP/2.0/TCP some.proxy.net:5078;branch=z9hG4bK32d3.45058b373afc3bd5897903a83bc0de7d.0 Via: SIP/2.0/UDP some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK32d3.257af0eab041dbd2a240416d1c6e454f.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPjc442ef42-b769-4fdc-81cf-33453aeaae67 From: "Rick" sip:1022@10.199.240.179;tag=b44068d2-d1af-4b56-87c0-2ea334784c94 To: sip:jyu3xsfkrz6c5qn@10.199.240.135 Contact: sip:asterisk@10.199.240.179:7060;alias=10.199.240.179~7060~1 Call-ID: f9f4006d-8df9-4501-a77d-d3f7ba45c327 CSeq: 14602 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Rick" sip:1022@10.199.240.179 Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 461
v=0 o=- 95944032 95944032 IN IP4 34.206.126.53 s=Asterisk c=IN IP4 34.206.126.53 t=0 0 m=audio 30082 RTP/AVP 0 101 a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:30083 a=ptime:20 a=ice-ufrag:qgZIRgJo a=ice-pwd:mT7rDx0qrjOkoC4Yzlodqfl7ma a=candidate:aUPmOZbVuTuCQ7B9 1 UDP 2130706431 34.206.126.53 30082 typ host a=candidate:aUPmOZbVuTuCQ7B9 2 UDP 2130706430 34.206.126.53 30083 typ host
but this one was succeed
2020/01/26 19:07:07.349559 37.17.41.5:25171 -> 10.199.240.19:5078 SIP/2.0 180 Ringing Via: SIP/2.0/TCP some.proxy.net:5078;branch=z9hG4bK32d3.45058b373afc3bd5897903a83bc0de7d.0 Via: SIP/2.0/UDP some.proxy.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK32d3.257af0eab041dbd2a240416d1c6e454f.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPjc442ef42-b769-4fdc-81cf-33453aeaae67 From: "Rick" sip:1022@10.199.240.179;tag=b44068d2-d1af-4b56-87c0-2ea334784c94 To: "1030_DEV" sip:jyu3xsfkrz6c5qn@10.199.240.135;tag=63F0AFCC-6C807D61 CSeq: 14602 INVITE Call-ID: f9f4006d-8df9-4501-a77d-d3f7ba45c327 Contact: sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp Record-Route: sip:34.202.130.141:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge, sip:34.202.130.141:5078;r2=on;lr;nat=yes;rtp=bridge, sip:3.220.42.184:5078;lr User-Agent: PolycomVVX-VVX_450-UA/5.8.0.13851 Allow-Events: conference,talk,hold Accept-Language: en Call-Info: sip:some.proxy.net;appearance-index=1 Content-Length: 0
-- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Thanks Alex. I found the issue! That was in device side
https://knowledgebase-iframe.polycom.com/kb/viewContent.do?externalId=15474
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Hello! Thanks for reply! Removing unnecessary headers didn't solve issue. Switching on TCP didn't solve it neither.
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Alex Balashov -
Kjeld Flarup -
Zhan Bazarov