Network path
Loss, jitter, latency, saturated links, and QoS.
VoIP voice quality
Poor or one-way audio in VoIP and Asterisk
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.
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
Delay, echo, dropouts, or one-way audio?
Describe the symptom, call direction, network, and timestamps. We will correlate the media path with signaling and system data.