Best tool for SIP stress testing

Chapter 1 SIPP IntroductionSIPP is a tool software used to test the performance of the SIP protocol. This is a GPL open source software.
It contains some basic sipstone user proxy workflows (UAC and UAS) and can be used to create and release multiple calls using invite and B ye. It can also read the XML scenario file, that is, the configuration file that describes any performance tests. It dynamically displays test running statistics (call rate, back-to-back signal latency, and message statistics ). Periodically dump the CSV statistics. TCP and UDP on multiple sockets can be reused by means of re-transmission management. You can use regular expressions in the scenario definition file to dynamically adjust the call rate.
SIPP can be used to test many real sip devices, such as sip proxy, b2buas, SIP Media Server, SIP/X gateway, and sip pbx, it can also simulate thousands of SIP proxy calls to your SIP System. There are a lot of information about SIPP found on Google, but there are few or many of them are incomplete installation instructions. Sipp url: Http://sipp.sourceforge.net/ SIPP: Http://sourceforge.net/project/showfiles.php? Group_id = 104305 & package_id = 119322(When I am using rc6, rc8 has come out. | -.-) Four SIPP installation methods:1) No TLS support or password verification support:

a)       # tar -xvf sipp-1.1rc6.tar.gz

b)       # cd sipp-1.1.rc6

c)       # make

The SIPP file produced by make is an executable file. You only need to use the scenario XML file and CSV file to perform the SIP test. 2) support TLS and password verification, however, pcap audio playback is not supported:

a)       # tar -xvf sipp-1.1rc6.tar.gz

b)       # cd sipp-1.1.rc6

c)       # make ossl

In this way, the compiled file is added to the TLS support for password verification. 3) pcap play is supported, but password verification is not supported: (pcap play means RTP speech can be performed, but no 407 auth verification is performed)

a)       # tar -xvf sipp-1.1rc6.tar.gz

b)       # cd sipp-1.1.rc6

C) # Make pcapplay4) Support for pcap sound file playback and support for password verification (support for 407 auth verification) a) # tar-xvf sipp-1.1rc6.tarb) # cd sipp-1.1.rc6c) # Make pcapplay_ossl latest message: after using the sipp-1.1rc6, if the pcap mode is used for packet playback, through the packet capture can not grasp the session message body. The session body of the SIP cannot be viewed during multiple attempts and modifications to the configuration file. Later updated to the sipp-1.1rc8, capture the packet can see the SIP session body, it seems that other users have found this bug. Chapter 2 Introduction to several major call processes of SIPExample 1: Call invite is paused and the call ends. After a calls B, AST returns the 100 tring and 180 ring messages, Ack messages are returned here, and then pause 10 seconds to send the bye message. The system returns the 200 OK message. | (1) Invite | --------------- à | (2) 100 (optional) | <--------------- | (3) 180 (optional) | <------------------- | (4) 200 | <----------------- | (5) ACK | ----------------- à | (6) pause | (7) Bye | ---------------> | (8) 200 | <----------------- | Example 2: Invite call, establish a connection, and then RTP with the rfc2833 DTMF. After several seconds of delay, send the bye message, and the other party returns 200 OK. Scenario file: uac_pcap.xml (original XML file) sipp uac remote | (1) Invite | ------------------> | (2) 100 (optional) | <------------------ | (3) 180 (optional) | <------------------ | (4) 200 | <---------------- | (5) ACK | ------------------> | (6) RTP send (8 s) | |========================>|||| (7) rfc2833 digit 1 | ========================>|||| (8) bye | ------------------> | (9) 200 | <--- --------------- | Example 3: SIPP is processed as a SIP server. Remote sipp uas | (1) Invite | -----------------> | (2) 180 | <----------------- | (3) 200 | <----------------- | (4) ACK | -----------------> | (5) pause | (6) Bye | ------------------ >|| (7) 200 | <------------------ | chapter 1, Example 4: After successful sip register, invite to AST, AST returns the 100, 180, or 403 Forbidden message, and SIPP sends ACK, after a delay of MS, SIPP sends bye, ast back 200 okregister ---------- --> 200 <---------- 200 <---------- invite ----------> 100 <---------- 180 <---------- 403 <------------ 200 <---------- ack ----------> [5000 MS] Bye ----------> 200 <------------