VoIP & Call Center

Voice & contact center

VoIP & Call Center Setup

Carrier-grade voice and a complete contact center, engineered end to end. SIP trunking, IP-PBX, IVR, ACD, and omnichannel routing tuned to the quality thresholds that keep every call clear and every caller answered.

Why it matters

Voice quality is a budget, not an accident.

A good call is the sum of a latency budget you have to respect. VoIP carries signaling over SIP (RFC 3261) and media over RTP (RFC 3550), and every hop, buffer, and codec spends part of your one-way delay. The ITU-T G.114 recommendation puts the target at 150 ms or less, with jitter under 30 ms and packet loss under 1 percent. We design the network, codecs, and QoS to stay inside that budget, then measure it, so the quality holds under real load.

  • Built on standards. SIP signaling, RTP media, and proven IP-PBX platforms like FreePBX, Asterisk, and 3CX.
  • Sized for concurrency. Capacity is measured in concurrent calls, with QoS and headroom so peaks do not degrade voice.
  • Lower phone bills. Moving from legacy lines to SIP typically cuts phone spend by 30 to 50 percent.
One-way latency budgettarget under 150 ms
050100150 ms Network + Jitter buf + Codec ~120 ms

Illustrative budget: network transit plus jitter buffer plus codec delay stays under the 150 ms G.114 target. Actual figures measured per deployment.

By the numbers

The thresholds that define clear calls.

0 ms
One-way latency target
0%
Voice uptime target
30-0%
Typical phone bill reduction
<0%
Packet-loss target
What we build

From dial tone to a full contact center.

A complete voice layer, from SIP trunks and IP-PBX to omnichannel routing – pick the scope that fits.

SIP trunking & IP-PBX

We provision SIP trunks and stand up an IP-PBX on FreePBX, Asterisk, or 3CX, sized for your peak concurrent calls. Signaling rides SIP (RFC 3261), media rides RTP (RFC 3550), and QoS keeps voice prioritized. Delivered on-prem, in the cloud, or hybrid.

SIP RFC 3261RTP RFC 3550FreePBX / Asterisk3CX

IVR & call flows

Menu design, self-service prompts, and business-hours routing that get callers to the right place fast.

ACD & skills-based routing

Automatic call distribution that matches each caller to the best available agent by skill and priority.

Omnichannel & CCaaS

One queue across voice, chat, email, and SMS, with CTI screen pops and WebRTC softphones. Run it in-house or on a CCaaS platform such as Genesys, Five9, Amazon Connect, or Twilio, and grow capacity in software.

Recording & QA

Call recording, quality-assurance scoring, and compliance controls to coach agents and meet obligations.

Real-time dashboards

Live queue, agent, and service-level metrics so supervisors can act while it still matters, not after.

Voice codecs

Choosing the right codec for the link.

Codec choice trades bandwidth against quality. Representative figures include IP overhead – we tune per circuit and QoS policy.

Voice codecs with bitrate, bandwidth need, and best use
CodecBitrateBandwidth per callBest use
G.71164 kbps~87 kbpsToll quality on LAN and high-bandwidth links
G.7298 kbps~31 kbpsBandwidth-constrained WAN, about 8x saving
Opus6 to 510 kbpsAdaptiveModern WebRTC and variable network conditions
Call quality thresholds with target and acceptable ranges
MetricIdeal targetAcceptableReference
One-way latency<= 150 ms150 to 400 msITU-T G.114
Jitter< 30 msBuffer-dependentVoice best practice
Packet loss< 1%Concealment to ~2%Voice best practice
How we deliver

From assessment to live calls.

01 / ASSESS

Discover

We baseline call volume, concurrency, number inventory, and network quality against the G.114 latency budget.

02 / DESIGN

Architect

Choose on-prem, cloud, or CCaaS; pick codecs, SIP trunks, IVR and ACD flows, and the QoS plan.

03 / BUILD

Provision & port

Stand up the IP-PBX or contact center, port your numbers, and run parallel routing during cutover.

04 / OPERATE

Monitor & tune

Measure latency, jitter, and loss in production, then tune routing and capacity as call patterns change.

Questions

VoIP & call center, answered.

Do we keep our existing phone numbers?
Yes. We port your existing numbers – local, toll-free, and international where supported – to the new platform through a managed local number portability process. We coordinate the port dates with the losing carrier so calls keep flowing, and we run parallel routing during cutover to avoid any gap in service.
What is the difference between a SIP trunk and a PRI?
A PRI is a legacy digital circuit with a fixed 23 usable voice channels per line over dedicated copper or fiber. A SIP trunk carries voice as packets over your IP connection, so capacity is measured in concurrent calls and scales in software rather than by adding physical lines. SIP is typically cheaper, faster to provision, and easier to burst, which is why most migrations move from PRI to SIP.
How much bandwidth does each call need?
It depends on the codec. G.711 delivers toll-quality voice at about 87 kbps per call including IP overhead, while G.729 compresses to roughly 31 kbps for about an eight-times bandwidth saving with a small quality trade-off. Opus scales between the two. We size the circuit for peak concurrent calls, add headroom, and apply QoS so voice is prioritized over other traffic.
Should we run on-premises or move to CCaaS?
On-premises IP-PBX platforms such as FreePBX, Asterisk, or 3CX give you full control and predictable costs, which suits fixed sites and strict data-residency needs. Cloud and CCaaS models trade capital for elasticity, built-in redundancy, and fast rollout of omnichannel features. We assess your call volume, agent locations, compliance needs, and existing hardware, then recommend on-prem, cloud, or a hybrid.
Can you build a full contact center with IVR and ACD?
Yes. We build complete contact centers with IVR menus, automatic call distribution, skills-based routing, and omnichannel queues across voice, chat, email, and SMS. That includes CTI screen pops, WebRTC softphones, call recording, quality assurance workflows, and real-time dashboards, delivered on-prem or on a CCaaS platform such as Genesys, Five9, Amazon Connect, or Twilio.
VoIP & Call Center Setup

Every call clear.
Every caller answered.

Tell us how many agents you run, where they sit, and what your numbers are today. We will assess your voice quality and propose a platform with SIP, routing, and a migration plan.