6 May
2024
6 May
'24
11:38 a.m.
Thanks Ben1
that did not make it!
Just to be clear, there is no UAS scenario for SIPp running. The call is
generated by SIPp scenario, in "UAC mode", to Kamailio and Kamailio will
reply back to the originating SIPp instance with a 300 Multiple Choices
message. Setting the "rrs" attribute in the "recv" element doesn't
work.
As for next_url, it did not help either....
*Sérgio Charrua*
On Mon, May 6, 2024 at 3:46 PM Ben Kaufman <bkaufman(a)bcmone.com> wrote:
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…),
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(a)lists.kamailio.org>
*Sent:* Monday, May 6, 2024 6:57 AM
*To:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
*Cc:* Sergio Charrua <sergio.charrua(a)voip.pt>
*Subject:* [SR-Users] Retransmission behaviour after ACK
*CAUTION:* This email originated from outside the organization. *Do not
click links or open attachments* unless you recognize the sender and know
the content is safe.
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 :
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword.
-->
<send retrans="500" start_rtd="true" rrs="true">
<![CDATA[
INVITE sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[call_number]@[remote_ip]:[remote_port]>
Contact: sip:sipp@[local_ip]:[local_port];transport=[transport]>
Call-ID: [call_id]
CSeq: 102 INVITE
User-Agent: SIPp
Supported: replaces, timer
Content-Type: application/sdp
Subject: Performance Test
Session-Expires: 3600;refresher=uas
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=SIPp
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100" rss="true"> </recv>
<recv response="300" rtd="true" rss="true">
<action>
<ereg regexp="\<([a-zA-Z0-9@.=;]+)[^>]+"
search_in="hdr"
header="Contact: "
assign_to="1"/>
<log message="INFO: 300 Contact Header: [$1]" />
<exec command="echo [$1] >> ./300_ContactHeader.txt" />
</action>
<action>
<ereg regexp="sip:.*@([0-9A-Za-z\.]+):([0-9]+);transport=([A-Z]+)"
search_in="hdr" header="Contact:" check_it="true"
assign_to="dummy,host,port,transport" />
</action>
</recv>
<!-- Reference variables="dummy" /-->
<!-- Send ACK for 180 Ringing
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 -->
<send rrs="true">
<![CDATA[
ACK sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:[call_number]@[remote_ip]:[remote_port]>[peer_tag_param]
Contact: <sip:sipp@[local_ip]:[local_port]>
Call-ID: [call_id]
CSeq: 102 ACK
User-Agent: SIPp
Content-Length: 0
]]>
</send>
<!-- definition of the response time repartition table (unit is ms) -->
<!-- ResponseTimeRepartition value="10, 50, 100, 150, 200, 500, 1000"/
-->
</scenario>
The question is : is there anything wrong with my kamailio script? or with
SIPp scenario? why does kamailio have different behaviours between Asterisk
and SIPp?
Any option to disable retransmissions on Kamailio ? (using UDP)
Thanks in advance,
*Sérgio Charrua*