[SR-Users] t_relay_to_udp - 500 server error occurred (6/SL)

Donat Zenichev donat.zenichev at gmail.com
Thu Nov 9 09:10:08 CET 2017


Hi all.

I have a problem with t_relay_to_udp function.

My topology:
UAC -> tcp -> kamailio <-> udp <-> asterisk

Kamailio processes requests from clients over TCP and relays requests to
asterisk over UDP.

Kamailio gets INVITE from client, relays it to asterisk, then he sends 100
trying to user client and abruptly sends 500 internal error to client
(after 100 trying).
But the main interesting thing is, that asterisk answers on INVITE and
dialog continues between kamailio and asterisk (without user agent client)
- it reaches the  destination then, without originator : )

How error on kamailio looks like:
Nov  8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: tm [t_fwd.c:1716]:
t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding
Nov  8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: sl [sl_funcs.c:363]:
sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server
error occurred (6/SL)

The same problem is, when I change transport protocol between kamailio and
asterisk to TCP,
using the t_relay_to_tcp function.
When I use t_relay function, it works fine, nothing happens that can drop
dialogs between kamailio and clients.

Relaying code part:
if ($var(to_asterisk)=="1") {
xlog("L_NOTICE","Relaying to asterisk - change socket to udp:10.1.1.1:5070
\n");

force_send_socket(udp:10.1.1.1:5070);

if (!t_relay_to_udp()) {
sl_reply_error();
}
}

I need to use t_relay_to_udp with force_send_socket, because kamailio
changes the source port (source socket) every time he generate new INVITEs
to asterisk and this is irrelevant.

Dump between kamailio and user agent client:

INVITE sip:1111111111example at kam1.domain.com:5070;transport=TCP SIP/2.0
Via: SIP/2.0/TCP
uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-
Max-Forwards: 70
Contact: <sip:debug.device-example at uac.domain.example:5065;transport=TCP>
To: <sip:1111111111example at kam1.domain.com:5070;transport=TCP>
From: <sip:debug.device-example at kam1.domain.com:5070
;transport=TCP>;tag=93394e63
Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
Proxy-Authorization: Digest username="debug.device-example",realm="
kam1.domain.com
",nonce="WgQGlVoEBWnhH3CF6DE77uKxKJx0tT4e",uri="sip:1111111111example at kam1.domain.com:5070
;transport=TCP",response="4753aa849715d26d0dde1b3c2314c843",a
orithm=MD5
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 241

v=0
o=Z 0 0 IN IP4 uac.domain.example
s=Z
c=IN IP4 uac.domain.example
t=0 0
m=audio 8000 RTP/AVP 8 3 110 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/TCP
uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486
To: <sip:1111111111example at kam1.domain.com:5070;transport=TCP>
From: <sip:debug.device-example at kam1.domain.com:5070
;transport=TCP>;tag=93394e63
Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM.
CSeq: 2 INVITE
Server: MS Lync
Content-Length: 0

SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL)
Via: SIP/2.0/TCP
uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486
To: <sip:1111111111example at kam1.domain.com:5070
;transport=TCP>;tag=9a78ee8ed9a3d4ba441c41d83ba2e904.4113
From: <sip:debug.device-example at kam1.domain.com:5070
;transport=TCP>;tag=93394e63
Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM.
CSeq: 2 INVITE
Server: example
Content-Length: 0


Thanks in advance for hints.




-- 
-- 
BR, Donat Zenichev
Wnet VoIP team
Tel:  +380(44) 5-900-800
http://wnet.ua
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