24 Jan
2020
24 Jan
'20
8:34 p.m.
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
25 Jan
25 Jan
10:20 a.m.
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-…
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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
26 Jan
26 Jan
3:23 a.m.
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-…
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(a)lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
10:15 p.m.
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
27 Jan
27 Jan
10:46 a.m.
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(a)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
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31 Jan
31 Jan
6:01 p.m.
Thanks Alex. I found the issue! That was in device side
https://knowledgebase-iframe.polycom.com/kb/viewContent.do?externalId=15474
--
Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
26 Jan
26 Jan
10:07 p.m.
Hello! Thanks for reply!
Removing unnecessary headers didn't solve issue.
Switching on TCP didn't solve it neither.
--
Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html
1759
days inactive
1766
days old
6 comments
3 participants
participants (3)
-
Alex Balashov -
Kjeld Flarup -
Zhan Bazarov