VoIP Toolbox now offers SIP monitoring, from basic SIP OPTIONS health checks to more advanced scenarios like registration or calling.
SIP Endpoint Monitoring is a service where VoIP Toolbox regularly monitors your endpoint with either SIP OPTIONS, SIP Registration or an actual SIP INVITE call scenario. It emails you an alert if the response is not what you expect. Typically that’s just a non 2xx final response code, but it might be something else for your specific setup.
For SIP OPTIONS monitoring that is over UDP or TCP there is no cost, anyone can sign up for a free account and use it. For TLS monitoring, or the SIP Registration or SIP INVITE Call scenarios there are more feature rich tiers available.
How does it work?
I think a future blog series could do a deep-dive into the infra setup - let me know if this would be of interest. For now, we can do a 60 second overview.
VoIP Toolbox uses a worker-based architecture.
High level SIP monitoring architecture
The interesting bit is perhaps those workers. They take SIP monitoring jobs and then effectively wrap around a SIPp instance.
SIPp is one of the de facto SIP testing tools. I have intentionally built the monitoring feature with SIPp as the underlying SIP driver, because it paves the way for more advanced test scenarios. For example, as a VoIP Toolbox user you could upload your own custom SIPp scenario file and have it alert when the run differs from what’s configured. This could work for either client or server files (i.e. for tests where you are making or receiving a call on your own infra).
VoIP Toolbox doesn’t quite go that far yet, but it’s very close. Let me know if that’s interesting and I can give you beta access to that feature.
(Potential) Upcoming Features
The SIP endpoint monitoring has been on my desired feature list pretty much since day one of building the VoIP Toolbox site. I am keen to hear from VoIP Professionals though on what would make this more useful. Some feedback and suggestions I’ve had already include:
- Custom scenarios (i.e. run any SIPp regularly and monitor the output). As mentioned above, some beta support is already available for this - let me know if you want to try it!
- Regular RTP Metrics Monitoring. Take the SIP INVITE call scenario, VoIP Toolbox would gather RTP stats like delay, jitter, number of dropped packets etc, and alert when the defined bounds are breached. This could also build valuable QoS data. e.g. Do your users receive noticeably poorer call audio quality in your peak times?
- User RTP Testing. Generate a link to send to a customer, VoIP Toolbox then carries out an RTP stream quality test, to capture live data on the user’s network quality. Perhaps most useful when combined with RTP monitoring above (i.e. you can show the user’s network is likely at fault, not your own).
- Notification on DNS changes.
- Improved PCAP and SIP log processing. The site already has some support here, but
there could be various improvements, e.g.
pcapngsupport, capability to extract SIP from flat log files etc.
Have an idea that should be on the list? Let me know.
Next Time…
Another feature on VoIP Toolbox that is already live is SIP GeoDNS Lookup. I’ll likely
blog about that next time, it was an interesting build. You should give it a
try. Any free tier account can access it. Try
a query for sip.pstnhub.microsoft.com for an interesting example.