[SR-Users] Bad request when INVITE relayed to another kamailio

Alex Balashov abalashov at evaristesys.com
Sat Jan 25 08:20:07 CET 2020


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-header-diet/

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 at 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 at 10.199.240.179>;tag=ba402508-a640-409f-ba30-dffdfe499f43
> To: <sip:jyu3xsfkrz6c5qn at 10.199.240.135>
> Contact: <sip:asterisk at 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 at 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 at 10.199.240.179>;tag=ba402508-a640-409f-ba30-dffdfe499f43
> To: <sip:jyu3xsfkrz6c5qn at 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 at 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/



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