What Is VoIP and How Can It Reduce Your Business Phone Bill by Up to 60%?

Every month, businesses across Bangladesh pay significant sums to BTCL, Grameenphone, Robi, and other carriers for outgoing telephone calls — local calls, inter-city calls, and international calls to clients, suppliers, and overseas offices. For an organisation with 50 staff making business calls throughout the day, the monthly telephone bill can easily reach BDT 2–5 lakh. Multiply that across a year, and you are looking at BDT 25–60 lakh spent on telephone calls.

VoIP — Voice over Internet Protocol — is the technology that can cut that bill by 40–60%, while simultaneously adding features that traditional telephone systems never offered. Understanding what VoIP is, how it works, and what it requires of your network gives you the information you need to make a sound decision about whether a VoIP upgrade makes financial sense for your organisation.

What Is VoIP?

VoIP is the technology of transmitting voice telephone calls as digital data packets over an Internet Protocol (IP) data network — the same type of network that carries your email, web browsing, and file transfers. Rather than establishing a dedicated circuit between two phones (as traditional PSTN telephony does), VoIP converts your voice into digital data, compresses it using an audio codec, packages it into IP packets, and transmits those packets across the data network to the destination, where they are reassembled and converted back to sound.

The destination can be another VoIP phone on the same network (an internal extension), a VoIP user on another network (reached via the internet), or a traditional telephone on the PSTN (reached via a SIP trunk — an IP-based gateway to the public telephone network).

SIP: The Protocol That Makes VoIP Work

SIP — Session Initiation Protocol — is the dominant signalling protocol for VoIP. It handles the "signalling" side of a VoIP call: the initial handshake that establishes the call, the negotiation of audio codec parameters, and the termination of the call when either party hangs up. SIP is analogous to the process of dialling a number on a traditional phone — it sets up the connection.

Once SIP has established the call, the actual voice audio travels using RTP — Real-time Transport Protocol. RTP packets carry the compressed voice data in real time between the two endpoints.

SIP is an open standard, which is why equipment from different vendors — a Yealink IP phone, a Fanvil desk set, a softphone app on a smartphone — can all work with a Grandstream UCM or Yeastar IP PBX. This interoperability is one of VoIP's key advantages over proprietary traditional PABX systems where handsets were typically locked to a single manufacturer.

Audio Codecs: How Voice Is Compressed

The audio codec determines the quality and bandwidth consumption of a VoIP call. Common codecs used in Bangladesh enterprise VoIP systems include:

• G.711: Uncompressed (or minimally compressed) audio at 64 kbps per call. Excellent voice quality. Standard for LAN calls between IP phones.
• G.729: Highly compressed audio at 8 kbps per call. Good voice quality for trunk calls over internet connections. Widely used for SIP trunk calls where bandwidth is a cost or capacity concern.
• G.722: Wideband (HD) audio at 64 kbps per call. Significantly better voice quality than G.711 — voice sounds natural and full-bodied. Supported by premium IP phones from Yealink, Fanvil, Poly, and Avaya.
• Opus: Modern codec with excellent quality at variable bitrates (6–510 kbps). Increasingly common for software clients and video conferencing integration.

How VoIP Saves Money: The Three Cost Streams

VoIP reduces your telephone expenditure through three distinct mechanisms.

1. Eliminating or Reducing Per-Minute PSTN Call Charges

Traditional PSTN calls from Bangladesh use carrier-charged minutes. BTCL's business tariff for local calls is approximately BDT 1.5–2.5 per minute. Inter-city calls run BDT 2.0–3.0 per minute. International calls — to the US, UK, Middle East — range from BDT 2 per minute (some VoIP-based international tariffs) to BDT 20+ per minute on traditional PSTN international rates.

VoIP over SIP trunks — where your IP PBX connects to a SIP trunk provider rather than PSTN — dramatically reduces per-minute costs:

• Local calls via SIP trunk: BDT 0.5–1.0 per minute (typical SIP trunk pricing in Bangladesh, via providers offering termination to BTCL and mobile networks)
• Inter-city calls via SIP trunk: BDT 0.8–1.5 per minute
• International calls via VoIP: BDT 0.5–3.0 per minute depending on destination, vs. BDT 8–25 per minute on traditional international PSTN rates

For an organisation making BDT 3 lakh per month in PSTN call charges, switching to SIP trunk calling at 50% lower per-minute rates saves BDT 1.5 lakh per month — BDT 18 lakh per year.

2. Free Internal Calls Between All Extensions

In a traditional PABX connected to multiple office locations — a head office and branch offices — inter-office calls typically travel via the PSTN and incur per-minute charges. An employee in Dhaka calling a colleague in Chittagong pays inter-city PSTN rates.

With VoIP, all calls between extensions on the same IP PBX system — regardless of location — travel as data over your IP WAN (Wide Area Network) or internet connection. They cost nothing in per-minute charges. For an organisation with branches in Dhaka, Chittagong, Sylhet, and Rajshahi, the savings on inter-office calling alone can be substantial: eliminating 1,000 inter-office call minutes per day at BDT 2 per minute saves BDT 60,000 per month — BDT 7.2 lakh per year.

3. Eliminating PSTN Line Rental Costs

Traditional PABX systems require PSTN trunk lines — typically one analog line (or ISDN channel) per simultaneous outgoing call. Each BTCL business analog line costs BDT 1,000–1,500 per month in line rental, plus per-minute call charges. An organisation with 20 trunk lines pays BDT 20,000–30,000 per month in line rental before making a single call.

VoIP SIP trunks replace analog lines. A SIP trunk carries multiple simultaneous calls over a single internet connection — the number of concurrent calls is a software-defined capacity parameter, not a physical line count. A single SIP trunk subscription might support 20 simultaneous calls for a monthly fee of BDT 5,000–8,000 — dramatically less than 20 individual PSTN line rentals.

Bangladesh PSTN vs. VoIP Cost Comparison

Let's build a concrete comparison for a 100-employee Bangladesh organisation:

Traditional PSTN setup:

• 10 PSTN trunk lines: BDT 15,000/month rental
• Local calls (2,000 min/day × 22 days × BDT 2.0): BDT 88,000/month
• International calls (200 min/day × 22 days × BDT 10): BDT 44,000/month
• Inter-office calls (Dhaka to Chittagong, 500 min/day × 22 days × BDT 2.5): BDT 27,500/month
• Total monthly: BDT 174,500 (BDT 1.74 lakh)

VoIP setup (IP PBX + SIP trunks):

• SIP trunk (10 channels): BDT 6,000/month
• Local calls via SIP (2,000 min/day × 22 days × BDT 0.8): BDT 35,200/month
• International calls via VoIP (200 min/day × 22 days × BDT 2): BDT 8,800/month
• Inter-office calls: BDT 0 (internal extension calls, free)
• Total monthly: BDT 50,000 (BDT 0.50 lakh)

Monthly saving: BDT 1.24 lakh (71% reduction) Annual saving: BDT 14.9 lakh

These figures are representative rather than precise — actual savings depend on your specific call volumes, destinations, and current carrier tariffs. But the direction and magnitude are consistent with what SSAL's clients experience after VoIP migration.

What Your Network Needs to Support VoIP

VoIP places specific requirements on the data network. Understanding these requirements allows you to assess whether your existing network can support VoIP without modification, or whether network upgrades are needed as part of the VoIP project.

Bandwidth

Each simultaneous VoIP call consumes:

• G.711 codec: approximately 87 kbps (including RTP packet headers and IP overhead)
• G.729 codec: approximately 31 kbps
• G.722 (wideband): approximately 87 kbps

For a 100-user organisation with a peak of 20 simultaneous calls, G.711 requires approximately 1.74 Mbps of network bandwidth. G.729 requires only 620 kbps. Modern enterprise internet connections (10 Mbps or higher from BTCL fiber or private ISPs) provide far more than adequate bandwidth for VoIP.

The internet bandwidth requirement for SIP trunks is similarly modest. 20 simultaneous external calls in G.729 require just 620 kbps of internet bandwidth — negligible compared to typical enterprise internet allocations.

Quality of Service (QoS)

The more important network requirement is Quality of Service (QoS) — the mechanism that prioritises voice packets over data packets in the network to prevent delays and jitter. Without QoS, a large file download or video stream can saturate a network link and cause voice packets to be delayed, resulting in voice quality degradation (chopped audio, echo, one-way audio).

Modern managed enterprise switches support QoS through DSCP (Differentiated Services Code Point) marking — tagging voice packets with a high-priority marking that tells all network devices to handle them before lower-priority traffic. IP phones automatically mark their traffic with the correct DSCP value. The IP PBX and network switches must be configured to honour this marking.

For most Bangladesh enterprise LANs with modern managed switches (Cisco, HP/Aruba, TP-Link Business), QoS configuration is a standard part of VoIP network preparation. It requires network administrator access and a few configuration changes — not a network hardware replacement.

Network Latency and Jitter

VoIP voice quality is sensitive to two network characteristics:

Latency (delay): Total one-way delay should be below 150ms for good voice quality. On a well-configured local LAN, latency between extension phones is typically 1–5ms — well within this limit. For SIP trunk calls across the internet, latency depends on routing and distance to the SIP provider's servers.

Jitter: Jitter is the variation in packet arrival time. VoIP endpoints use a de-jitter buffer to smooth out packet timing variation, but excessive jitter (above 30ms) exhausts the buffer and causes audio artifacts. A well-configured QoS policy on the enterprise LAN minimises jitter for internal calls.

The VoIP Implementation Process

A successful VoIP migration follows a structured process:

Phase 1 — Network Assessment: Evaluate existing LAN switches for QoS capability. Measure current internet bandwidth and confirm it is adequate. Identify any network segments (old unmanaged switches, long cable runs) that need upgrading before VoIP deployment.

Phase 2 — IP PBX Design: Select the IP PBX platform appropriate for your user count, feature requirements, and budget. Design the extension plan, ring groups, IVR menus, call queues, and voicemail configuration.

Phase 3 — SIP Trunk Procurement: Select a Bangladesh SIP trunk provider and negotiate capacity and per-minute rates. Number porting (transferring your existing PSTN numbers to the SIP provider) requires BTCL coordination and typically takes 2–4 weeks.

Phase 4 — Installation and Configuration: Install the IP PBX hardware, configure SIP trunks, configure IP phones, and set up the dial plan. Test all internal and external call scenarios before cutover.

Phase 5 — Cutover and Training: Schedule the cutover during off-peak hours (typically weekend night). Brief all staff on using the new system — particularly IP phone features like call transfer, conferencing, and voicemail access. Retain analog PSTN lines as a fallback for the first 30 days.

Phase 6 — Optimisation: Review call quality metrics and adjust QoS settings. Monitor SIP trunk utilisation to confirm capacity is adequate. Address any reported call quality issues.

Is VoIP Right for Your Organisation?

VoIP delivers clear financial savings for virtually every Bangladesh enterprise with:

• More than 20 staff making regular business calls
• Any inter-office calling between multiple locations
• Regular international calling
• Existing managed network switches and reliable internet connectivity

The financial case is strongest for organisations with high inter-office call volumes (eliminating those charges entirely) and regular international calling (where the per-minute cost differential is most dramatic).

Get Free Consultation — Contact SSAL to receive a VoIP cost savings analysis based on your actual current telephone bill. We review your current PSTN expenditure, model the VoIP cost structure, and show you exactly how much you can save — before you commit to any investment.