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Automate your OpenSIPS config tests - OpenSIPS Summit 2024

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Andreas Granig

These are the slides of my OpenSIPS Summit 2024 presentation about automating your test calls. It dives into why automated call testing is crucial, how to integrate it into your CI/CD pipeline and how to extend testing of single calls into load testing and testing of other protocols. The presentation also provides details how open source components such as sipp, asterisk and rtpengine are used to implement agents to generate test calls for SIP, Fax and WebRTC at scale.

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Automating your OpenSIPS config tests Andreas Granig <agranig@sipfront.com>
It¡¯s me, Andreas
¡°Let¡¯s build our own test framework ¡­¡±
¡°... but I already run Homer!¡±
Issues happen.
Time for some (Github) Action!
Github CI Trigger sipfront-app command & control baresip SIP UA kamailio SIP proxy rtpengine media handler sipfront-persistor Kafka to DB TimestreamDB metrics PostgreSQL events sipfront-finalizer condition evaluator System under Test MQTT SIP RTP/RTCP stats via MQTT/Kafka HTTPS SQL SIP RTP/RTCP Github GH action via Sipfront API
Agent Anatomy System under Test Orchestration Aggregation generate terminate kamailio rtpengine kamailio rtpengine sipp baresip codecept kamailio rtpengine Docker asterisk
Types of Tests ¡ñ SIP ? SIP calls ¡ñ Mobile App ? SIP calls ¡ñ Browser WebRTC ? SIP calls ¡ñ FAX ¡ñ ¡­
T-38 is fantastic
T-38 is fantastic Northrop T-38 Talon
T-38 is fantastic Northrop T-38 Talon T.38 Fax
Surprises! ¡ñ Offer/Answer handling (Re-INVITE, 415, 488) ¡ñ Transcoding on SBCs
Fax Setup System under Test Orchestration Aggregation sipfront-agent kamailio rtpengine asterisk AMI internal external SIP UDP IP4 RTP IP4 G711/T.38 SIP UDP/TCP/TLS IP4/6 RTP SDES/DTLS IP4/6 G711/T.38 FAX_MODE_T38 FAX_MODE_G711 FAX_SUCCESSFUL FAX_FAILED
App/WebRTC ? SIP ¡ñ Many variations in the wild ¡ð OS, Browser ¡ð IPv4/v6 ¡ð RTP/SRTP ¡ð G711/Opus ¡ñ Device Farms help!
Lifting the carpet
Lifting the carpet
WebRTC Setup Web App sipfront-agent appium Browserstack codeceptjs HTTPS WebRTC Orchestration Aggregation Audio recordings SDP offer/answer RTCP statistics to SIP agent on B side
SIP ? SIP ¡ñ Basic Call + Media at scale ¡ñ Presence ¡ñ ¡°Exotic¡± call flows ¡ð E911 + multipart/mime ¡ð SCAIP/NowIP Telecare ¡ñ IMS?
SIP Setup System under Test sipfront-agent kamailio rtpengine sipp ctrl_udp internal external SIP UDP IP4 play-media SIP UDP/TCP/TLS IP4/6 RTP SDES/DTLS IP4/6 Orchestration Aggregation Post Dial Delay CLI Info Retransmits Loss RTT Jitter Loss Packets/Bytes in/out
¡°The boy who cried wolf¡±
Key Takeaways & Learnings ¡ñ Test early & often ¡ñ Keep usage barrier low ¡ñ Make it reliable
THANK YOU Andreas Granig <agranig@sipfront.com>

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Automate your OpenSIPS config tests - OpenSIPS Summit 2024