SIPp’s syntax for handling record route and setting ACK request lines correctly kinda sucks.
Try setting `rrs` in your `recv` element (https://sipp.readthedocs.io/en/latest/scenarios/ownscenarios.html?highlight=rrs#list-of-commands-with-their-attributes),
and then use `next_url` as the contact in your ACK: (https://sipp.readthedocs.io/en/latest/scenarios/keywords.html#next-url)
Kaufman
From: Sergio Charrua via sr-users <sr-users@lists.kamailio.org>
Sent: Monday, May 6, 2024 6:57 AM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Cc: Sergio Charrua <sergio.charrua@voip.pt>
Subject: [SR-Users] Retransmission behaviour after ACK
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Hi all!
I'm testing some redirect scenarios with Kamailio 5.7.4 and using Astterisk 13 and SIPp for unit testing.
The kamailio's script is simple and uses TM, HTTP_ASYNC_MODULE, JSONRPC, JANSSON, RTJSON and XLog modules.
Request route is simple:
request_route {
route(HANDLE_OPTIONS);
route(HANDLE_RETRANS);
route(HANDLE_CANCEL);
route(REQINIT);
route(HANDLE_INVITE);
}
where HANDLE_RETRANS is:
route[HANDLE_RETRANS]{
# handle retransmissions
xlog("L_INFO","HANDLE_RETRANS - message type is $mt for method $rm");
if (!is_method("ACK")) {
if(t_precheck_trans()) {
xlog("L_INFO","HANDLE_RETRANS - Precheck Trans");
t_check_trans();
xlog("L_INFO", "HANDLE_RETRANS - Exiting after Retransmission check - method $rm");
exit;
}
xlog("L_INFO","HANDLE_RETRANS - Check Trans");
t_check_trans();
}
}
and HANDLE INVITE is:
route[HANDLE_INVITE]{
if (is_method("INVITE"))
{
xlog("L_INFO","MAIN - calling TO_CARRIER");
setflag(FLT_ACC);
route(TO_CARRIER);
exit;
}
}
The HTTP requests call a REST API returning a JSON object, that is iterated and produces a CONTACT header containing multiple choices using the append_branch() command.
The RELAY route is :
route[RELAY] {
# Sends a 300 Multiple Choices back to the proxy that requested the routing lookup
xlog("L_INFO","REPLY_302 - send reply");
send_reply("300", "Multiple Choices");
xlog("L_INFO","REPLY_302 - exit");
exit;
}
Which returns indeed a Contact header with multiple contact choices.
When using Asterisk to originate a call to Kamailio, the SIP call flow is as follows and terminates (SNGrep and TCPDump do not detect anything else):
SRC DST
INVITE (SDP)
──────────────────────────>
100 trying -- your call is
<──────────────────────────
300 Multiple Choices
<──────────────────────────
ACK
──────────────────────────>
However when using SIPp I get some retransmissions after the final ACK.
SRC DST
INVITE (SDP)
──────────────────────────>
100 trying -- your call is
<──────────────────────────
300 Multiple Choices
<──────────────────────────
ACK
──────────────────────────>
300 Multiple Choices
<<<────────────────────────
300 Multiple Choices
<<<────────────────────────
300 Multiple Choices
<<<────────────────────────
SIPp scenario is :