To be a bit more specific, the issue would appear to be that Kamailio is not correlating the ACK with the transaction. Compare the ACK sent by Asterisk to the ACK sent by SIPp to see what is wrong with the SIPp ACK.
From: Sergio Charrua <[email protected]> Sent: Monday, May 6, 2024 10:39 AM To: Ben Kaufman <[email protected]> Cc: Kamailio (SER) - Users Mailing List <[email protected]> Subject: Re: [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. 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 <[email protected]<mailto:[email protected]>> 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=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 <[email protected]<mailto:[email protected]>> Sent: Monday, May 6, 2024 6:57 AM To: Kamailio (SER) - Users Mailing List <[email protected]<mailto:[email protected]>> Cc: Sergio Charrua <[email protected]<mailto:[email protected]>> 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]+)<sip:.*@([0-9A-Za-z\.%5d+):(%5b0-9%5d+);transport=(%5bA-Z%5d+)>" 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
__________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to [email protected] Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
