VoIP voice quality

We find where voice quality breaks

We measure RTP, packet loss, jitter, codecs, NAT, and load. We fix the cause instead of masking it with buffers.

  • RTP
  • Jitter
  • Codecs
RTPthe actual media stream
QoSpriority and network path
NATtwo-way reachability

What we inspect

We inspect the entire voice path

The network, server, carrier, and agent endpoint all shape one call.

Network path

Loss, jitter, latency, saturated links, and QoS.

RTP and NAT

SDP addresses, ports, asymmetric routing, and session timeouts.

Codecs and resources

Transcoding, mismatched settings, CPU, and endpoints.

Outcome

Cause, correction, control

After the fix, there is a way to detect degradation before complaints spread.

Measure

We compare healthy and failing calls using evidence.

Targeted change

We fix the specific path, rule, codec, or configuration.

Observe

We monitor loss, registrations, resources, and RTP/SIP errors.

FAQ

Frequently asked questions

Short answers before work begins.

Why is audio present in only one direction?

Common causes are RTP addressing, NAT, or firewall rules. The exact one appears when SIP/SDP is correlated with traffic.

Will changing the codec help?

Sometimes. It will not fix packet loss or a wrong route, so measurement comes first.

Can quality be monitored continuously?

Yes. Network, registrations, resources, and call indicators can reveal degradation early.

Next step

Start with a focused assessment

Describe the outcome you need and leave the best contact for a reply.

Discuss your project