Asterisk + Kamailio + RTPengine Deployment
A production voice platform usually means Kamailio or OpenSIPS at the edge, RTPengine for media and Asterisk for application logic. Zypager builds the full stack together — routing, media relay, dialplan and trunks — installed, integrated, tested and documented so your team can run and extend it.
Problems solved
- No clean reference architecture for the full stack
- Layers that do not integrate cleanly
- One-way audio across the assembled system
- Undocumented builds nobody can change safely
How Zypager helps
- Design and install edge, media and application layers
- Integrate signalling and media end to end
- Configure routing, trunks and basic hardening
- Document the build so it is reproducible
Technical areas covered
Typical deliverables
- Installed, integrated and tested voice stack
- Commented configuration across all layers
- Architecture & run documentation
- Next steps for HA and scale
Frequently asked questions
Yes. One-way and no-audio problems are usually in the RTP media path or NAT/SDP handling rather than the signalling. With a SIP trace and the SDP from each leg, the media direction can be followed to the point where it stops.
Yes — Asterisk, FreeSWITCH, Kamailio, OpenSIPS and RTPengine. Many real systems combine them (edge proxy, media relay and application logic), and problems often live at the boundaries between them.
WebRTC-related work may be reviewed separately where it is directly connected to Asterisk, Kamailio, RTPengine or SIP troubleshooting, but it is not currently offered as a standard fixed-price service.
Reading traces is central to how Zypager works. A clear pcap, sngrep or Homer export, plus call details, is often enough to identify the failing transaction without live access.
Yes, on an hourly basis for urgent incidents — failed inbound/outbound calls, RTP/no-audio and carrier problems. Final scope and availability depend on access and complexity.
Related services
Kamailio Consulting
SIP edge routing, registrar and dispatcher built to scale.
RTPengine Consulting
Media relay and SRTP/RTP bridging that actually carries audio.
Asterisk Consulting
Dialplan, PJSIP and ARI that are easier to debug and safer to run.
Not sure which service you need?
Send the symptoms and a call example. The right starting point becomes obvious from the trace.