Hi Everyone, i am new to a testing tool called SIPp and i am trying to do stress testing for asterisk, i am running this command:
./sipp -sn uac -d 20000 -s 44444 {asterisk server IP address} -l 100 -r 10 -trace_err -error_file sipperror
But i am getting error on my asterisk CLI stating that no matching endpoint is found, please help if anyone familiar with this type of error:
[Apr 7 02:50:24] NOTICE[59495]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘INVITE’ from ‘“sipp” sip:sipp@127.0.1.1’ failed for ‘asterisk ip:5061’ (callid: 604-59494@127.0.1.1) - No matching endpoint found
The endpoint is reachable on asterisk server and to make sure i checked pjsip show endpoints and can confirm there is a sip registration to that endpoint, Here is my pjsip file if it is related:
All PJSIP traffic gets associated with an endpoint. Your request from SIPp did not match an endpoint, so it was rejected and challenged for authentication.
You’d need to add another endpoint for SIPp, if named “sipp” then it would be matched as that is the username in the “From” header.
Hey jcolp, I added another endpoint called “sipp” in my asterisk configuration files but i am still getting the same error. Do i have to change anything in my sipp command?
Okay i restarted my asterisk and i stopped getting the previous error on asterisk CLI which was the “No matching endpoint found”, but i am still getting the error on SIPp in my error file
You haven’t specified what error occurred, shown console output, or described what is happening in SIPp so I can’t answer. I’d suggest taking a step back, breaking down each bit, figuring out what you want to happen, and then piecing it together and figuring out when something doesn’t work what exactly about it isn’t working.
If you are talking about this error them make sure you use sipp with SIP user authentication because your SIP endpoints in pjsip are configured to use user authentication.
As you know i will have to provide “-ap” in my sipp command for password authentication, in addition, the scenario should have auth=“true” in the . So my first step was to create my own XML scenario but when i run the sipp command i get:
Unable to load or parse ‘playback_with_initial_sdp.xml’ xml scenario file.
I was not getting this error before because i was using a default scenario but it will not work in my case due to authentication, can i solve this issue by removing the user authentication in my pjsip.conf file on asterisk?