This is a collection of SIPp scenarios for testing purposes.
SIPp - download and documentation: http://sipp.sourceforge.net
Thanks: https://github.com/saghul/sipp-scenarios
Name | Description |
---|---|
sipp_uac_audio_video.xml |
UAC audio-video |
sipp_uac_basic.xml |
UAC basic |
sipp_uac_pcap_g711a.xml |
UAC audio PCM-A 8000 (G.711) |
sipp_uas_basic.xml |
UAS basic |
sipp_uas_delayed_answer.xml |
UAS delayed answer |
sipp_uas_pcap_g711a.xml |
UAS audio PCM-A 8000 (G.711) |
Server side:
sipp -sf sipp_uas_basic.xml -i <server address> -p 5060
Client side:
sipp -sf sipp_uac_basic.xml -m 1 <server address>:5060
Server side:
sipp -sf sipp_uas_pcap_g711a.xml -i <server address> -mi <server address> -mp 6000
Client side:
sipp -sf sipp_uac_pcap_g711a.xml -m 1 -i <client address> -mi <client address> -mp 6000 <server address>:5060
Server side:
sipp -sf sipp_uas_486_busy.xml -i <server address> -p 5060
Client side:
sipp -sf sipp_uac_basic.xml -m 1 <server address>:5060
Server side:
sipp -sf sipp_uas_no_answer.xml -i <server address> -p 5060
Client side:
sipp -sf sipp_uac_cancel.xml -m 1 <server address>:5060
Server side:
sipp -sf sipp_uas_403_forbidden -i <server address> -p 5060
Client side:
sipp -sf sipp_uac_basic.xml -m 1 <server address>:5060
It is possible simulate a multi-party call using sipp_uac_pcap_g711a.xml
and sipp_uas_pcap_g711a.xml
scenario files. In this example, Alice and Bob both call Charlie. In a real use case, Charlie is supposed to mix his own stream with the Alice one to send it to Bob in a single RTP audio session. In the same way, Charlie is supposed to mix his own stream with the Bob on to send it to Alice.
Charlie side:
sipp -sf sipp_uas_pcap_g711a.xml -i <Charlie's address> -mi <Charlie's address> -mp 6000
Bob side:
sipp -sf sipp_uac_pcap_g711a.xml -m 1 -i <Bob's address> -mi <Bob's address> -mp 6000 <Charlie's address>:5060
Alice side:
sipp -sf sipp_uac_pcap_g711a.xml -m 1 -i <Alice's address> -mi <Alice's address> -mp 6000 <Charlie's address>:5060