SIP Trunking Explained: Connecting Your IP PBX to the Public Phone Network in Bangladesh

You have invested in an IP PBX — or you are planning to. You understand that it will handle internal calls between your extensions efficiently. But how does it connect to the outside world? How do your employees call customers on BTCL landlines or Grameenphone mobile numbers? How do clients dial your main number and reach your receptionist?

The answer is SIP trunking — and understanding how it works is essential to planning a complete, cost-effective enterprise telephony system in Bangladesh.

What Is SIP Trunking?

A SIP trunk is a virtual telephone line that connects your IP PBX to the Public Switched Telephone Network (PSTN) via an internet or private data connection, using the Session Initiation Protocol (SIP). It is the VoIP replacement for the physical analog telephone lines or ISDN circuits that traditionally connected a PABX to the telephone network.

The word "trunk" comes from traditional telephony, where trunk lines are the shared lines connecting a PABX to the carrier network — used by multiple extensions for external calls. A SIP trunk serves the same function as those physical lines, but:

It carries calls as IP data rather than analog signals
It runs over your existing internet connection (or a dedicated IP connection)
It supports a defined number of simultaneous calls (channels) rather than requiring one physical line per call
It costs significantly less than equivalent physical PSTN lines

Think of a SIP trunk as a motorway with a defined number of lanes. Each lane carries one simultaneous call. If you have a 20-channel SIP trunk, you can have 20 calls in progress simultaneously — some incoming, some outgoing, distributed across all your extensions.

How SIP Trunks Replace ISDN/PSTN Lines in Bangladesh

In Bangladesh's traditional enterprise telephony model, organisations connected their PABX to the public network using:

Analog PSTN lines from BTCL: Each physical line supports one simultaneous call. Costs approximately BDT 1,000–1,500 per month per line in rental, plus per-minute call charges.
E1 ISDN lines: A single E1 circuit carries 30 simultaneous voice channels over a physical connection. Used by larger enterprises requiring higher trunk capacity. Monthly E1 line rental from BTCL is approximately BDT 25,000–40,000.

SIP trunks replace these physical circuits. Instead of a physical line from BTCL's exchange to your PABX, you configure your IP PBX with the SIP credentials (SIP server address, username, password, authentication parameters) provided by your SIP trunk provider. Your IP PBX then registers with the provider's SIP server over the internet and routes outgoing calls through that provider. Incoming calls from the PSTN are received by the provider and delivered to your IP PBX via SIP.

SIP Trunk Providers in Bangladesh

Bangladesh's SIP trunk market is developing but still maturing compared to markets like India or Southeast Asia.

Grameenphone Enterprise

Grameenphone (GP) Enterprise offers SIP trunk services to enterprise customers as part of its enterprise voice portfolio. GP's nationwide GSTN (mobile network) coverage makes it practical for terminating calls to both BTCL landlines and all Bangladeshi mobile networks. GP Enterprise SIP trunks are typically sold with dedicated IP circuits rather than over public internet.

Robi Enterprise

Robi Axiata's enterprise division offers SIP-based voice services for corporate customers, often bundled with data connectivity products. Robi's coverage and pricing are competitive with Grameenphone for enterprise voice applications.

Local VoIP Providers and Resellers

Several local technology providers in Bangladesh resell SIP trunk capacity from upstream carriers or international VoIP wholesale providers. These resellers may offer competitive per-minute rates for international calls in particular, using wholesale VoIP routes that bypass expensive traditional international PSTN rates.

When evaluating local SIP providers, assess:

BTRC licensing: The provider should hold a valid BTRC (Bangladesh Telecommunication Regulatory Commission) licence for VoIP services. Unlicensed VoIP termination is technically illegal in Bangladesh and carries legal risk for your organisation.
Network quality: Request SIP trunk trial periods and evaluate call quality with actual traffic before committing to a contract.
Support responsiveness: SIP trunk issues (wrong caller ID, one-way audio, registration failures) require prompt technical support to resolve. Evaluate the provider's support availability and response times.
Redundancy: Does the provider operate redundant SIP servers? What is their Service Level Agreement (SLA) for uptime?

Capacity Planning: How Many SIP Channels Do You Need?

Getting SIP trunk channel capacity right is important. Too few channels and external callers get busy signals during peak periods. Too many channels and you are paying for capacity you never use.

Calculating Your Required Channel Count

Start with your Busy Hour Call Attempt (BHCA) data — if you have call records from your existing PSTN system, extract the busiest hour of the busiest day and count the maximum simultaneous external calls during that period. This is your peak concurrent call requirement.

If you do not have call records, use this rule of thumb: for a general office environment, plan approximately 1 simultaneous external call for every 4–6 employees. For call centre environments, plan 1 channel per agent position.

For a 100-employee organisation: 100 ÷ 5 = 20 simultaneous channels. Add 20% headroom: 24 channels. Round up to 25 channels as a practical specification.

For a 20-seat inbound call centre: 20 agent positions + 5 overflow channels = 25 channels minimum.

Inbound vs. Outbound Channel Allocation

Most SIP trunk providers offer pooled channel counts — any call, inbound or outbound, uses one channel from your pool. This pooling is more efficient than the traditional model where inbound and outbound lines were separate physical circuits.

Monitor actual concurrent call data during the first 3 months of SIP trunk operation and adjust channel count if you consistently approach capacity (indicating under-provisioning) or rarely use more than 50% of your channels (indicating over-provisioning).

DID Numbers and Number Porting

DID (Direct Inward Dialing) allows callers to reach specific extensions or departments within your organisation by dialling a specific number directly — without going through a receptionist. In a traditional PABX with DID, the carrier provides a range of numbers (say, 02-9999200 to 02-9999299) that all route to your PABX, which then connects the call to the appropriate extension based on the last 2 digits dialled.

SIP trunks support DID natively. Your SIP provider allocates a range of DID numbers to your SIP trunk. When an incoming call arrives for any DID number in your range, the SIP trunk delivers it to your IP PBX with the called number in the SIP headers. Your IP PBX routes the call to the appropriate extension or department queue based on its dial plan configuration.

Number Porting from PSTN to SIP

If you have existing BTCL numbers that you want to retain when migrating to SIP trunking, you can port (transfer) those numbers to your SIP trunk provider. The number porting process in Bangladesh:

Request porting from your SIP provider: The provider submits a porting request to BTCL on your behalf.
BTCL verification: BTCL verifies that you are the authorised subscriber for the number and that your BTCL account is clear of outstanding balances.
Porting execution: On the agreed porting date, the number transfers from BTCL routing to the SIP provider routing. Your existing callers continue to reach you at the same number, now via SIP trunk.

The number porting process typically takes 2–4 weeks in Bangladesh and requires coordination between your organisation, your SIP provider, and BTCL. Plan this timeline into your VoIP migration project.

Disaster Recovery: Geographic Number Flexibility

One of SIP trunking's most valuable but least appreciated advantages is geographic number flexibility for disaster recovery. Traditional PSTN numbers are physically tied to a specific BTCL exchange and location. If your office is inaccessible due to flooding (a real risk in Dhaka), fire, or other disaster, your PSTN numbers cannot be re-routed to an alternate location quickly — BTCL number re-routing typically takes days.

With SIP trunking, your DID numbers are software-assigned to your SIP trunk account. Your SIP provider can re-route incoming calls to any SIP endpoint or any PSTN number within minutes — redirecting your Dhaka office's incoming calls to a Chittagong branch, to a temporary office, or to a mobile phone number — simply by changing the routing table in their management portal.

This capability significantly enhances business continuity. As part of your organisation's Business Continuity Plan (BCP), document the procedure for emergency SIP trunk re-routing so that your IT or communications team can execute it quickly when needed.

QoS Configuration for Voice Quality

Voice quality on SIP trunks depends on the quality of your internet connection and the QoS configuration of your network equipment. Even with adequate bandwidth, poor QoS configuration can result in choppy audio, echo, or one-way audio during periods when other internet traffic (large file downloads, backup transfers, video streaming) competes for bandwidth.

The Core QoS Principle for SIP Trunks

Voice packets must be prioritised over data packets at every point where congestion could occur — primarily at the internet-facing router/firewall interface. The router should be configured to:

Mark or preserve DSCP values: Voice packets are marked with DSCP EF (Expedited Forwarding, decimal 46) by the IP PBX and IP phones. The router must honour this marking and forward EF-marked packets with the highest priority.

Queue voice traffic in the high-priority queue: Configure a strict priority queue for EF-marked traffic on the internet-facing interface.

Limit voice bandwidth allocation: While voice gets priority, ensure that a single large file download cannot completely starve voice traffic — configure fair queuing for other traffic classes.

Consider a dedicated internet circuit for SIP trunks: For large deployments or organisations where voice quality is critical, a separate internet connection (even a modest 10 Mbps circuit) dedicated to SIP trunk traffic eliminates the competition with general internet traffic entirely.

Codec Selection for SIP Trunks

G.729 is the recommended codec for SIP trunk external calls — its 8 kbps per call bandwidth is extremely efficient, consuming minimal internet bandwidth while delivering acceptable voice quality. G.711, while better sounding, uses 8x more bandwidth and is better suited to LAN calls between internal extensions.

Configure your IP PBX to use G.729 (or G.711 as a fallback if G.729 is unavailable) for SIP trunk calls, and G.711 or G.722 for internal LAN extension calls.

Implementing SIP Trunking: The Practical Checklist

Before your SIP trunk goes live, confirm all of the following:

IP PBX is configured with SIP trunk provider credentials and registration is confirmed
Outbound dial plan correctly routes calls via SIP trunk for external numbers
Inbound DID routing correctly directs incoming calls to the right extensions/queues
Emergency call routing: 999 (or local emergency numbers) reaches emergency services — SIP providers must support emergency calling in Bangladesh
Caller ID presentation: your organisation's main number appears correctly on outgoing calls
G.729 codec licence installed on IP PBX (G.729 requires a licence on many IP PBX platforms)
QoS is configured on the internet-facing router
SIP ALG (Application Layer Gateway) is disabled on the router/firewall — SIP ALG on consumer and some enterprise routers interferes with SIP signalling
Failover to analog PSTN backup line is tested and working
BTCL number porting is complete and incoming calls are verified

Getting Your SIP Trunk Right from Day One

SIP trunking is the right choice for virtually every Bangladesh enterprise deploying an IP PBX. The cost savings over PSTN line rental and per-minute call charges are compelling, and the disaster recovery flexibility is a genuine business continuity enhancement.

The key to a successful SIP trunk deployment is selecting a reliable provider, configuring QoS correctly, and validating all call scenarios before going live. SSAL's unified communications team handles all of these elements as part of our IP PBX deployment projects.

Talk to Our Expert — Contact SSAL to discuss SIP trunk options for your organisation. We work with multiple Bangladesh-licensed SIP providers and can recommend the right provider for your call volumes, quality requirements, and connectivity environment.