01
Freezing video and robotic audio
Packet loss and buffer underruns on mobile networks.
Packet loss, echo, jitter and lag in WebRTC calls
WebRTC video and audio quality optimization
Make video calls hold up on real networks: SFUs close to users, simulcast, TURN relays for restrictive firewalls, and quality metrics you can watch.
Symptoms
Signs your platform has this problem
If several of these sound familiar, the plan below is where we would start.
02
Calls failing behind strict firewalls
ICE negotiation failing because TURN relays are missing or misconfigured.
03
CPU and battery drain
Phones decoding high-resolution streams they can't display.
Remediation plan
How we fix it, step by step
Each phase ends with a measurement, so you can see what changed before the next one starts.
01
SFU media servers
Selective forwarding units (for example LiveKit) deployed close to your users.
02
Simulcast and bitrate adaptation
Sending several quality layers and switching each receiver to the one its network can handle.
03
TURN relays
TURN servers over TLS on port 443, so calls get through restrictive corporate and hospital networks.
04
Codecs and echo cancellation
Hardware-accelerated VP8, H.264 or AV1 encoding, and acoustic echo cancellation.
Technical checklist
Remediation checklist
What we check before a change goes to production:
- Run SFU media servers in the regions your users are in
- Enable simulcast with quality switching based on receiver bandwidth
- Run TURN relays over TLS on port 443 for restrictive networks
- Collect WebRTC quality stats (jitter, packet loss, RTT) in your monitoring
What we measure
We take a baseline first and report the same measurements after each change, from your own monitoring — evidence, not promised results.
- Connect rate
- Calls that connect, split by network type
- Round-trip time
- RTT, jitter and packet loss from WebRTC stats
- Call quality
- Freezes and audio drops per call minute
Related service
Custom software
Custom software for Indian SMEs and startups, built around your workflow with an agreed scope, review milestones, clear ownership and documented handover.
Explore Custom softwareQuestions
Questions about this remediation
A selective forwarding unit receives each participant's streams and forwards the right quality layer to each receiver, without re-encoding.
With TURN servers that relay media over TLS on port 443, which most firewalls allow even when they block the UDP ports WebRTC prefers.
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Want an engineer to look at this with you?
Send us the symptoms and any metrics you have. We'll reply within one business day, set up a call and agree what to measure before anything changes.