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Latency in a VoIP phone call has nothing to do with the control protocols and everything to do with your connectivity and the configuration of the jitter buffer
by wolrah 7y ago
Latency in a VoIP phone call has nothing to do with the control protocols and everything to do with your connectivity and the configuration of the jitter buffers on either end.
There should be no practical difference in audio performance, either latency or quality, between different VoIP signaling protocols unless something is wrong. If you're experiencing latency on calls that is significantly greater than your network latency plus jitter then something is wither misconfigured or malfunctioning.
That said as RIPT attaches the media to the signaling in a single flow of packets it should have far fewer problems negotiating NATs and similar scenarios that have been known to cause problems for SIP. Those problems cause complete loss of audio in one or both directions though, anything affecting call quality is almost certainly a network performance issue.
- Aloha 7y agoCall setup times with SIP are often much longer than their TDM predecessors, its one of my complaints with SIP.
- bcrl 7y agoUsually that means your ATA is not configured properly. With the correct dial plan settings, call setup time should be comparable to the PSTN. I have many other complaints with SIP, mostly related to the assumption by many clients that there is never any packet loss during call setup, but call setup time is not one.
- anon102010 7y agoThe idea that RIPT will be phone endpoint to phone endpoint for media is ridiculous which means latency will go up. I hope I misread the spec! Because phones are usually double natted / double firewalled the vonages / ringcentral and friends will have the phones connect to their servers, and pass the media that way. I currently finally got the media side to handoff pretty directly from our phone system to whatever media endpoint is suggested to handle the call termination for SIP - latency is improved, but the media bridges to local telco etc seem to be adding their own latency. What is crazy is you used to be able to call three floors down (separate network but next door) and get no latency. Literally. The call was circuit switched through something in town or sometimes closer. Now with RIPT that media is going to travel to the ringcentral west point server, then back, to go three floors, and YES, latency will be higher than 30 year old tech. Imagine that. Billions spent, and the quality is worse. I hope this is wrong.
- wolrah 7y agoAs you've noted, the direct media thing is great but doesn't really work reliably with phones behind NAT unless there's special processing at the NAT side. And of course most NAT devices that offer said processing get it wrong and usually make things worse. I've been working in VoIP for years. We do direct media when we can, but we bounce it through the server fairly often because NAT sucks. RIPT seems to built primarily with cloud hosted applications in mind, which need to scale and are likely being connected to by clients behind NAT. It's explicitly not like SIP, where every user agent is equal and would theoretically communicate directly where possible. This is why I wish IPv6 had a real killer app to make it catch on, it'd make life so much easier for me and SIP would work how it's supposed to.