VoIPBin Documentation
Trunk
Overview
VoIPBIN’s Trunk system provides the infrastructure for SIP-based voice communication, supporting both outbound calls (trunking) and inbound calls (registration). Trunks connect VoIPBIN to the PSTN and enable SIP endpoints to communicate with external networks.
The Trunk system provides:
SIP trunking for outbound call routing
Registration services for inbound call delivery
Connection between VoIPBIN and external providers
Integration with on-premise PBX systems
Flexible routing through providers
How Trunks Work
The trunk system handles bidirectional call flow between VoIPBIN and external networks.
Trunk Architecture
+-----------------------------------------------------------------------+
| Trunk System |
+-----------------------------------------------------------------------+
+-------------------+
| VoIPBIN |
| Platform |
+--------+----------+
|
+-------------------+-------------------+
| |
v v
+-------------------+ +-------------------+
| Trunking | | Registration |
| (Outbound Calls) | | (Inbound Calls) |
+--------+----------+ +--------+----------+
| |
v v
+-------------------+ +-------------------+
| Providers | | SIP Endpoints |
| (Telnyx, Twilio) | | (Phones, PBX) |
+--------+----------+ +--------+----------+
| |
v v
+-------------------+ +-------------------+
| PSTN | | Registered |
| (Phone Network) | | Devices |
+-------------------+ +-------------------+
Key Components
Trunk: Connection to external SIP provider for outbound calls
Registration: Service for SIP endpoints to receive inbound calls
Provider: External SIP service (Telnyx, Twilio, etc.)
PSTN: Public Switched Telephone Network
Trunking vs Registration
Understanding the difference between these two services is essential.
Comparison
+-----------------------------------------------------------------------+
| Trunking vs Registration |
+-----------------------------------------------------------------------+
TRUNKING (Outbound):
+--------------------------------------------+
| Purpose: Make calls to external numbers |
| Direction: VoIPBIN -> Provider -> PSTN |
| Uses: Provider configuration |
| Example: Call +1-555-123-4567 |
+--------------------------------------------+
REGISTRATION (Inbound):
+--------------------------------------------+
| Purpose: Receive calls on SIP devices |
| Direction: PSTN -> VoIPBIN -> Device |
| Uses: Extension configuration |
| Example: IP phone receives incoming call |
+--------------------------------------------+
When to Use Each
Capability |
Trunking |
Registration |
|---|---|---|
Direction |
Outbound calls |
Inbound calls |
Configuration |
Provider + Route |
Extension |
Endpoint |
External phone number |
SIP device/softphone |
Authentication |
Provider credentials |
Extension credentials |
Outbound Call Flow (Trunking)
Outbound calls route through providers to reach external numbers.
Trunking Flow
VoIPBIN Provider PSTN
| | |
| 1. Initiate call | |
| to +1-555-1234 | |
+---+ | |
| | |
v | |
+-------+ | |
| Route | | |
| Engine| | |
+---+---+ | |
| | |
| 2. Select provider | |
| based on route | |
| | |
| 3. INVITE | |
+------------------------>| |
| | |
| | 4. Route to PSTN |
| +----------------------------->|
| | |
| | 5. 180 Ringing |
| |<----------------------------+
| 6. Ringing | |
|<------------------------+ |
| | |
| | 7. 200 OK (Answer) |
| |<----------------------------+
| 8. Connected | |
|<------------------------+ |
| | |
| Media (RTP) <-------------------------------> |
Inbound Call Flow (Registration)
Inbound calls reach registered SIP devices.
Registration Flow
PSTN VoIPBIN SIP Device
| | |
| 1. Call to VoIPBIN number | |
+----------------------------->| |
| | |
| | 2. Flow execution |
| | (dial extension) |
| +---+ |
| | | |
| | v |
| | +-------+ |
| | |Lookup | |
| | |Reg. | |
| | +---+---+ |
| | | |
| | 3. INVITE to registered IP |
| +----------------------------->|
| | |
| | 4. 180 Ringing |
| |<----------------------------+
| 5. Ringback | |
|<----------------------------+ |
| | |
| | 6. 200 OK |
| |<----------------------------+
| 7. Connected | |
|<----------------------------+ |
| | |
| Media (RTP) <-------------------------------------> |
Trunk Configuration
Configure the trunk system through providers, routes, and extensions.
Provider Configuration (for Trunking)
See Provider Overview for detailed configuration.
$ curl -X POST 'https://api.voipbin.net/v1.0/providers?token=<token>' \
--header 'Content-Type: application/json' \
--data '{
"name": "Primary SIP Provider",
"type": "sip",
"hostname": "sip.provider.com"
}'
Route Configuration (for Trunking)
See Route Overview for detailed configuration.
$ curl -X POST 'https://api.voipbin.net/v1.0/routes?token=<token>' \
--header 'Content-Type: application/json' \
--data '{
"name": "Default Outbound Route",
"provider_id": "provider-uuid-123",
"priority": 1
}'
Extension Configuration (for Registration)
See Extension Overview for detailed configuration.
$ curl -X POST 'https://api.voipbin.net/v1.0/extensions?token=<token>' \
--header 'Content-Type: application/json' \
--data '{
"name": "office-phone",
"username": "office1",
"password": "secure-password"
}'
Complete Setup Example
Set up a complete trunk system for bidirectional calling.
Setup Steps
Step 1: Configure Provider (Outbound)
+--------------------------------------------+
| Provider: "Telnyx Production" |
| Hostname: sip.telnyx.com |
| Type: sip |
+--------------------------------------------+
|
v
Step 2: Configure Route (Outbound)
+--------------------------------------------+
| Route: "Default Route" |
| Provider: Telnyx Production |
| Priority: 1 |
+--------------------------------------------+
|
v
Step 3: Create Extensions (Inbound)
+--------------------------------------------+
| Extension: "office-main" |
| Username: office-main |
| Password: ******** |
+--------------------------------------------+
|
v
Step 4: Register SIP Devices
+--------------------------------------------+
| Device registers to: |
| office-main@{id}.registrar.voipbin.net |
+--------------------------------------------+
|
v
Step 5: Configure Number Flow
+--------------------------------------------+
| Number: +1-555-123-4567 |
| Flow: Dial extension "office-main" |
+--------------------------------------------+
Result:
- Outbound: Make calls via Telnyx
- Inbound: Calls to +1-555-123-4567 ring office phone
Common Scenarios
Scenario 1: Basic Office Phone System
Set up a simple office with outbound and inbound calling.
Configuration:
+--------------------------------------------+
| Provider: Telnyx (for outbound) |
| Route: Default -> Telnyx |
| Extensions: office-1, office-2, office-3 |
| Number: +1-555-OFFICE |
+--------------------------------------------+
Call Flows:
+--------------------------------------------+
| Outbound: Any extension can dial out |
| -> Routes through Telnyx to PSTN |
| |
| Inbound: Calls to +1-555-OFFICE |
| -> IVR: "Press 1 for Sales..." |
| -> Routes to appropriate extension |
+--------------------------------------------+
Scenario 2: Multi-Provider Failover
Configure redundant providers for reliability.
Provider Configuration:
+--------------------------------------------+
| Primary: Telnyx |
| Route Priority: 1 |
| |
| Secondary: Twilio |
| Route Priority: 2 |
| |
| Tertiary: Bandwidth |
| Route Priority: 3 |
+--------------------------------------------+
Failover Behavior:
+--------------------------------------------+
| 1. Try Telnyx first |
| 2. If Telnyx fails (5xx) -> try Twilio |
| 3. If Twilio fails -> try Bandwidth |
| 4. All fail -> return error to caller |
+--------------------------------------------+
Scenario 3: Remote Worker Setup
Enable remote workers with softphones.
Setup:
+--------------------------------------------+
| Extension per worker: |
| - remote-john |
| - remote-jane |
| - remote-bob |
+--------------------------------------------+
Worker Configuration:
+--------------------------------------------+
| Softphone App: Zoiper, Linphone, etc. |
| Server: registrar.voipbin.net |
| Username: remote-john |
| Domain: {customer-id}.registrar.voipbin.net|
+--------------------------------------------+
Benefits:
+--------------------------------------------+
| - Work from anywhere with internet |
| - Same extension travels with worker |
| - Company number for outbound caller ID |
| - Inbound calls reach worker globally |
+--------------------------------------------+
Best Practices
1. Provider Management
Configure at least two providers for failover
Test provider connectivity regularly
Monitor call success rates per provider
Choose providers with coverage in your regions
2. Route Configuration
Set up priority-based routing
Configure failover for all routes
Test failover scenarios periodically
Document routing logic
3. Extension Security
Use strong, unique passwords
Rotate credentials periodically
Monitor for unauthorized registrations
Enable TLS when possible
4. Monitoring
Track call success/failure rates
Monitor provider latency and quality
Set up alerts for registration failures
Review call logs regularly
Troubleshooting
Outbound (Trunking) Issues
Symptom |
Solution |
|---|---|
Calls not connecting |
Check provider is configured; verify route exists; confirm provider credentials |
Wrong provider used |
Review route priorities; check for destination-specific routes |
All providers failing |
Check provider status; verify network connectivity; test each provider independently |
Inbound (Registration) Issues
Symptom |
Solution |
|---|---|
Device not receiving calls |
Verify registration is active; check extension in flow; confirm device is online |
Registration failing |
Check credentials; verify registrar address; confirm firewall allows SIP traffic |
Intermittent registration |
Check NAT settings; enable keep-alives; increase registration expiry |
Audio Issues
Symptom |
Solution |
|---|---|
No audio |
Check RTP ports; verify NAT traversal; confirm media IP is reachable |
One-way audio |
Enable STUN/TURN; check symmetric RTP; verify both endpoints can send/receive |
Poor audio quality |
Check network bandwidth; reduce jitter; consider QoS configuration |
Trunking
SIP trunking is a technology that enables organizations to make telephone calls over the internet, rather than traditional phone lines. Instead of using physical phone lines to connect to the public switched telephone network (PSTN), SIP trunking uses an internet connection to carry voice and other communications. SIP trunking is cost-effective, scalable and provides a range of features and benefits, such as better call quality, enhanced disaster recovery capabilities, and the ability to easily manage multiple locations or remote workers. Additionally, SIP trunking allows organizations to consolidate their voice and data networks, reducing the complexity and cost of their telecommunications infrastructure.
Server address
Once you created trunk, the voipbin generates the trunk server address for you.
sip:{your voipbin trunk domain name}.trunk.voipbin.net
Authentication
Currently, The VoIPBIN’s trunking authentication supports only the Basic authentication.
Basic authentication
IP-based authentication (coming soon)
Basic authentication
To make a SIP outgoing call through a VoIPBIN using basic authentication, you need to follow a few steps:
Choose a SIP client: You can use a software-based SIP client, such as Zoiper or X-Lite, or a hardware-based SIP phone, such as a Cisco or Grandstream phone.
Configure your SIP client: You need to configure your SIP client with VoIPBIN credentials, such as your name, extension, password, domain info.
Set up your outgoing call settings: In your SIP client, you need to specify the destination address(phone number or extension) you want to call and set any additional options, such as the call type, call quality, or call duration.
Initiate the call: Once you have configured your SIP client and set up your outgoing call settings, you can initiate the call by clicking on the call button or using a keypad command.
Authenticate your credentials: When you initiate the call, your SIP client sends your authentication credentials to the VoIPBIN, using the basic authentication method. The VoIPBIN then verifies your credentials and authorizes the call.
Make the call: Once your credentials are verified, the VoIPBIN establishes the call and connects you with the destination address.
By following these steps, you can make a SIP outgoing call through VoIPBIN using basic authentication. This process can be used for a variety of business and personal applications, such as remote work, conferencing, and customer support.
UA VoIPBIN Destination
| | |
|---------------- INVITE --------------->| |
|<-- 407 Proxy Authentication Required --| |
|---------------- ACK ------------------>| |
| | |
|----- INVITE with Authorization ------->| |
| |---------------- INVITE --------------->|
| | |
| |<--------------- 100 Trying ------------|
|<------------- 100 Trying --------------| |
| | |
| |<--------------- 180 Ringing -----------|
|<------------- 180 Ringing -------------| |
| | |
| |<---------------- 200 OK ---------------|
|<------------- 200 OK ------------------| |
|-------------- ACK -------------------->| |
| |----------------- ACK ----------------->|
| | |
| | |
|-------------- BYE -------------------->| |
| |----------------- BYE ----------------->|
| |<---------------- 200 OK ---------------|
|<------------- 200 OK ------------------| |
Call handle
Unlike VoIPBIN’s normal call handling, the VoIPBIN handles trunking outbound calls in a different way. The VoIPBIN executes special flow for the trunking call. It executes the follow features:
Enable the early media.
Relay the hangup cause.
Early media handle
The VoIPBIN enables the early-media feature for the trunking outbound call.
UA VoIPBIN Destination
| | |
===================================================================================
|----- INVITE with Authorization ------->| |
| |---------------- INVITE --------------->|
| |<--------------- 100 Trying ------------|
|<------------- 100 Trying --------------| |
| |<--------------- 183 Ringing -----------|
|<------------- 183 Ringing -------------| |
|<------------- RTP Media ---------------|<---------------- RTP Media ------------|
Relay hangup cause
The VoIPBIN delivers the hangup cause code from the outgoing call.
UA VoIPBIN Destination
| | |
===================================================================================
|----- INVITE with Authorization ------->| |
| |---------------- INVITE --------------->|
| |<--------------- 100 Trying ------------|
|<------------- 100 Trying --------------| |
| |<--------------- 404 NOT FOUND ---------|
|<------------- 404 NOT FOUND -----------| |
Domain name
The SIP Domain resource in VoIPBIN entails a personalized DNS hostname designed to accept SIP (Session Initiation Protocol) traffic for your account. When a SIP request is directed to this domain, like sip:alice@example.trunk.voipbin.net, it is directed to VoIPBIN. Subsequently, VoIPBIN verifies the request’s authenticity and channels it to the designated voice_url linked with the SIP domain.
This pivotal component facilitates the management of SIP traffic within your VoIPBIN account. It accommodates incoming SIP requests from diverse sources, ensuring seamless communication and integration with VoIPBIN services. Businesses and developers can leverage the SIP Domain resource to create bespoke DNS hostnames, seamlessly integrate VoIPBIN services into existing systems, and construct scalable, reliable SIP-based communication solutions. This capability is particularly beneficial for organizations seeking to manage SIP-based communications securely and efficiently.