Anyone seeing new types of latency / audio issues with AI voice workloads?
Curious to get perspectives from folks on the telecom / SIP side.
I’ve been looking at a number of voice AI setups recently (Twilio / SIP / WebRTC),
and what’s interesting is:
- demos usually look fine
- but once in production, latency and audio issues start to show up
Things like:
- inconsistent end-to-end latency
- occasional one-way audio
- jitter affecting real-time interaction
- call quality varying depending on routing path
From a telecom perspective, this raises a few questions:
- how much of this is just classic RTP / routing / QoS issues?
- vs something new introduced by AI workloads (e.g. bursty traffic, bidirectional streaming)?
Would be really interested to hear:
- are you seeing different traffic patterns from these AI voice systems?
- any best practices emerging for handling latency-sensitive AI calls?
Feels like a mix of old problems + new usage patterns.