25 Jan
2020
25 Jan
'20
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
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--
Alex Balashov | Principal | Evariste Systems LLC
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Web: http://www.evaristesys.com/, http://www.csrpswitch.com/