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PBXact - The basics of VoIP

As we start this Sangoma PBXact deployment series, I thought it might be helpful to start with some basic principles of VoIP. Here are a few terms to familiarize yourself with, defined in my own simplified way for the beginner to understand.
  • Latency = time to reach a destination
  • Jitter = variation in latency
  • Bandwidth = size of internet pipe to push traffic
  • SIP = start of negotiations between two endpoints that want to talk
  • RTP = actual voice traffic after SIP completes
  • SIP trunking = connects you to other SIP or PSTN networks
  • ChanSIP = stable, older, limited SIP protocol
  • PJSIP = newer, future, more features SIP protocol
  • Codecs = conversion & compression of your voice into digital
To expound on the above:
  • DSL generally has more jitter than Cable ISPs, thus worse for VoIP ( haven't had too many issues myself with the few CenturyLink DSL deployments I've had, but just something to be aware of) I use for this purpose since it gives me all this information when surveying a potential clients internet quality.
  • Some SIP providers carry the RTP traffic on the same session that the SIP session opened, while others use separate voice data sessions and thus require ports 10000-20000 to be forwarded to the PBX for RTP. 
  • Some some providers use the term "trunks" differently. SipStation treats a trunk like a single line. Technically, they allow one trunk a concurrent outbound and inbound call. (concurrency bursting allows a per minute "burst" past the trunk limit. Helpful for the occasionally busy time where the client receives more calls that they have purchased trunks for. Need to monitor this as it might be beneficial to bump up the trunk quantity)
  • ChanSIP vs PJSIP...just read the forums to find people on both sides shouting which is better. PBXact, as of 2019, defaults to ChanSIP. I usually leave the extension like that, haven't had any issues. The one time I did change a single extension to PJSIP, was when that specific extension needed two endpoints to register simultaneously to it so that both would ring. ChanSIP, as far as I could tell didn't support this. PJSIP worked great in this application.
  • Some codecs require more bandwidth but sound better. Depends on the environment. I believe the default in PBXact is G.711. To be honest, I've never had to change this.
I hope this was helpful. Please let me know if there is anything else I can answer or add to this post!

Sangoma PBXact deployment series post #2 will be coming soon! (I will link it in this post)

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