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https://github.com/jchavanton/voip_perf
SIP performance test tool
https://github.com/jchavanton/voip_perf
performance-testing sip
Last synced: 24 days ago
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SIP performance test tool
- Host: GitHub
- URL: https://github.com/jchavanton/voip_perf
- Owner: jchavanton
- Created: 2017-07-21T16:26:35.000Z (over 7 years ago)
- Default Branch: master
- Last Pushed: 2022-09-29T16:25:14.000Z (about 2 years ago)
- Last Synced: 2024-04-21T17:03:08.300Z (7 months ago)
- Topics: performance-testing, sip
- Language: C
- Homepage:
- Size: 213 KB
- Stars: 37
- Watchers: 7
- Forks: 13
- Open Issues: 0
-
Metadata Files:
- Readme: README.md
Awesome Lists containing this project
README
[](https://hub.docker.com/r/jchavanton/voip_perf/)
### ABOUT
VoIP perf
It is a SIP signaling performance testing application that can provide a server and a client.
It can produce latency stats at specified intervals in csv to measure the response time of any SIP endpoint.
It can also read a json configuration file and randomize numbers.
### Docker quick start
[quick start with docker](QUICK_START.md)
### Linux Debian building from sources
[see commands in Dockerfile](docker/Dockerfile)
### Example : starting a server
```bash
./voip_perf \
--local-port=5072 \ # server listening port
--trying \ # when receiving invite 100 trying will be send
--ringing \ # when receiving invite 183 will be sent
--thread-count=4 \ # number of server threads created
--delay=1500 # ringing delay in MS 1.5 second
```
### Example : starting a client
This will send one INVITE to a randomise number starting with +1206?????? to server 1.1.1.1
```bash
./voip_perf \
"sip:[email protected]" \ # target R-URI, > will be replaced by random digit
--method="INVITE" \ # method
--local-port=5072 \ # source port
--caller-id="+1??????????" \ # user part of the From header, > will be replaced by random digit
--count=1 \ # total calls to send
--proxy=2.2.2.2 \ # this will send the SIP message to a SIP proxy instead of the host in R-URI
--duration=5 \ # send BYE after 5 seconds
--call-per-second=500 \ # send 500cps
--window=100000 \ # maximum amount of in progress calls
--thread-count=1 \ # number of thread used
--interval=1 \ # reporting interval, everyone second a line is added to voip_perf_stats.log with latency metrics
--timeout 7200 # total run time , if all the requests where not send, voip_perf will stop and report scnenario timeout
# --verbose # runwith verbose logs
```
### Example : starting a client with TLS
```bash
./voip_perf "sip:[email protected]:5061;transport=tls" \
--local-port=5050 \
--method="INVITE" \
--caller-id="+1??????????" \
--interval=1 \
--count=20 \
--call-per-second=10 \
--window=25 \
--thread-count=1 \
--use-tls \
--timeout 7200 \
--tls-cert="tls/certificate.pem" \
--tls-key="tls/key.pem" \
--tls-calist="tls/ca_list.pem" \
--conf="conf.json"
# --verbose \
```
### Example : config in json file
```json
{
"server" : [
{
"response" : [
{ "code" : 200,
"reason" : "OK",
"probability" : 0.6
},
{ "code" : 404,
"reason" : "Not found",
"probability" : 0.3
},
{ "code" : 0,
"reason" : "drop/no response timeout",
"probability" : 0.1
}
]
}
],
"client": [
{
"extra-headers": {
"Foo": "Bar",
"Bar": "Foo"
}
},
{
"users": [
{
"ruri": "sip:[email protected]",
"extra-headers": {
"Bar": "Open"
}
},
{
"ruri": "sip:[email protected]",
"extra-headers": {
"Bar": "Close"
}
},
{
"ruri": "sip:[email protected]",
"extra-headers": {
"VIP": "True",
"Bar": "Open"
}
}
]
}
]
}
```
### Example : adding headers from a C header file
You can add custom headers in this file
```
include/custom_headers.h
```
### Additional notes
voip_perf was initially a fork from pjsip-apps/src/samples/pjsip-perf.c
http://www.pjsip.org/release/0.5.4/PJSIP-Dev-Guide.pdf