Quels sont les meilleurs moyens de surveiller et d’optimiser les performances SIP pour les entreprises multisites ?

Truthfully, monitoring SIP trunking is best monitored at the server end of the architecture. You will always want to ensure that that your MOS is kept tidy, and watch for excessive latency or jitter, but the true quality monitoring will take place at a reputable SIP provider. If you purchase SIP trunks from one of the far too many companies selling them out of a garage, or the back office of a funeral parlor, that's trouble. However quality providers will monitor the service without asking for assistance in the effort. Ditch the consistent pinging, all that will do for SIP is eat up bandwidth, then focus on circuit throughput and stability. Pair the quality trunks with a stable circuit and your worries should be minimal.

Quality of Service (QoS) is a method to prioritize network traffic going through a router to provide acceptable service to most users. QoS is especially important for Voice over IP (VoIP) applications, such as SIP, that rely on low latency and jitter for good call quality. There are different ways to determine whether there’s any sort of compromise in QoS for any SIP based devices, depending on the network configuration and the tools available. Some possible methods are: Using a network monitoring tool that can measure and report on key QoS metrics, such as packet loss, delay, jitter, and bandwidth utilization. Some examples of such tools are SolarWinds VoIP & Network Quality Manager, PRTG Network Monitor, and Wireshark.

The best way to determine whether there's any sort of compromise in QoS for any SIP based devices is to use a SIP monitoring tool that shows the basic networking metrics like latency, jitter and packet loss. SIP monitoring tools are useful in helping you isolate where the actual issue lies with the SIP based device in question. Usually, vendors provide their own SIP monitoring tools, but there's also third-party ones such as Wireshark.

Use encryption: Encryption can prevent eavesdropping and tampering of SIP messages and voice data. You can use secure protocols such as TLS (Transport Layer Security) or SRTP (Secure Real-time Transport Protocol) to encrypt your SIP traffic. Use authentication: Authentication can verify the identity of the SIP endpoints and prevent unauthorized access. You can use methods such as passwords, certificates, or tokens to authenticate your SIP users and devices. Use firewalls and SBCs: Firewalls and SBCs (Session Border Controllers) can filter and control the SIP traffic between your network and the Internet. They can block malicious or unwanted traffic, enforce policies, and provide NAT (Network Address Translation) and QoS.

Choosing the right SIP provider and codec depends on your business needs, such as voice quality, bandwidth, and compatibility. Here are some factors to consider: Voice quality: The best codec for VoIP quality is the G.722 codec, which has a Mean Opinion Score (MOS) of 5.0, indicating excellent voice quality. However, this codec may not be supported by all SIP providers or devices, so you may need to use a different codec, such as G.711, which has a MOS of 4.1 and is widely compatible2. G.711 also offers uncompressed high quality voice, but consumes more bandwidth than G.7223. Bandwidth: The amount of bandwidth you need for SIP trunking depends on the number of concurrent calls and the codec you use.