Overview

15 min

This article provides a complete overview of how enterprises can integrate with Exotel’s SIP Trunking (SIP). It includes configuration options (TCP, TLS, FQDN), onboarding steps, best practices, and supported use cases.

1. What is Exotel SIP Trunking (SIP)?

Exotel SIP Trunking allows enterprise PBX/SBC systems to connect directly with Exotel’s voice infrastructure using SIP over TCP or TLS. It supports both inbound and outbound PSTN calling through IP connectivity, with optional FQDN-based flexibility.

2. Supported Use Cases

Use Case	Description
Outbound SIP (IP → PSTN)	Calls from customer SIP infra routed to Exotel → PSTN
Inbound SIP (PSTN → IP)	Incoming calls to Exophone routed to customer SIP server
IP–PSTN Intermix	Bi-directional call routing between IP infra and Exotel VNs
SIP Bot Integration	Direct SIP-to-Bot integration with no agent involvement
FQDN-based Load Balanced SIP	FQDNs resolve dynamically to multiple IPs for cloud/hybrid setups
Native SIP Voicebot Integration	Direct SIP call to customer-hosted SIP-native voicebot platform

3. Transport Options

Transport	Port	Encryption	DNS Support	Use Case
TCP	5070	No	Yes	Default, legacy infra
TLS	443	Yes	Yes	Encrypted SIP + SRTP flows
FQDN	Any	TCP/TLS	Required	Cloud, autoscaling, HA SIP infra

4. Integration Path – High-Level Flow

What is Supported

SIP Trunking with TCP and TLS transport

SIP trunk routing using static IPs or FQDN

Bi-directional call support (IP to PSTN and PSTN to IP)

Exophones (Virtual Numbers) mapped to SIP trunks

SIP-to-Flow integration using Connect or IVR — See: SIP-to-Flow Integration Guide

SIP-to-SIP integration for native voicebot routing — See: Native SIP Voicebot Integration Guide

Note – vSIP Throttling

Exotel enforces a default vSIP rate-limit of 20 calls per minute (CPM) per trunk to safeguard carrier capacity and call quality.

If your traffic profile requires a higher burst rate, raise a request via your CSM or Support ticket. The capacity-planning team will review historical traffic, carrier limits, and QoS requirements and can increase the throttling threshold accordingly.

What is Not Supported

SIP over UDP (not supported)

SIP registration-based authentication

SIP traffic without prior account upgrade/KYC

Supported Regional Exophones for MUM SIP Setup

Veeno KA, DL, Mum, AP, GJ

Getting Started with Exotel vSIP (Veeno Accounts)

Create your Exotel account via my.in.exotel.com

Complete KYC and upgrade the account with help from your Exotel Account Manager: link

Procure Exophones by emailing [email protected] (mention region-specific need)

Decide trunking mode: TCP or TLS, Static IP or FQDN

Collect and share with Exotel:

Account SID

Exophones

Trunk source IPs (for outbound trunking)

Trunk destination IPs/FQDN (for inbound trunking)

Chosen transport protocol (TCP/TLS)

Once provisioned, your SIP traffic will be routed via Exotel’s regional edge PoP.

KYC Verification based on the setup required

Convert to Full Account – Exotel team upgrades it

Account SID

Exophones (required for PSTN origination)

Source IP(s) (for outbound)

Destination IP(s) or FQDN (for inbound)

Transport type (TCP/TLS)

Provisioning:

VOIP and VOIP-PSTN enabled

SIP trunk created in backend by Tech Support

Dial Whom URI format configured

Testing:

Inbound and Outbound calls validated

SIP packets reviewed using tools like sngrep

5. Configuration Types

a. TCP Trunking (Port 5070) — See: Exotel Exotel SIP Trunking TCP Integration Guide

Uses format: sip:<number>@pstn.in4.exotel.com:5070;transport=tcp

Requires static IP whitelisting

b. TLS Trunking (Port 443) — See: Exotel Exotel SIP Trunking TLS Integration Guide

Uses format: sip:<number>@pstn.in4.exotel.com:443;transport=tls

Encrypted SIP signaling and SRTP media

Recommended for secure communication

c. FQDN-based Trunking — See: Exotel Exotel SIP Trunking FQDN Integration Guide

No IP whitelisting required

Uses DNS lookup to resolve SIP server IP dynamically

Ideal for cloud or redundant infrastructure

6. FQDN Configuration Guidance

If you are using a DNS-resolvable FQDN instead of a static IP for your SIP server, please ensure the following:

Your SIP server is accessible via a public FQDN (e.g., sip.customer.com)

It resolves to an IP address reachable by Exotel

Specify the port (e.g., 5070 for TCP or 443 for TLS) and transport protocol

Share the FQDN, port, and transport protocol with your Exotel Account Manager

Exotel will configure the trunk to use your FQDN. This enables dynamic IP handling and is ideal for cloud-hosted or HA infrastructures.

Once this is provisioned, you can begin testing:

Map your VN to the SIP trunk and initiate a test call

Use tools like sngrep or tcpdump to validate INVITE requests and SRTP media flow

In the Dial Whom field, use format: sip:<number>@<fqdn>:<port>;transport=tcp|tls

Confirm the P-Asserted-Identity header shows the correct Leg1 number (caller identity)

7. Best Practices for vSIP Integration

Use FQDN if infra is cloud-based, HA, or load-balanced

Keep DNS TTL between 30–60 seconds for fast recovery

Avoid SIP ALG in NAT/firewall appliances

Use PCMA (G.711 A-law) as primary codec

Use TLS + SRTP for security-sensitive traffic

Validate trunk reachability before mapping VNs

Restrict RTP media port range to 10000–20000 (10K ports only)

Each SIP call uses 2 ports

Exotel media servers support 3000 concurrent calls → 6000 ports

A 50% buffer ensures capacity during retries, hold-ups, or port conflicts

Helps avoid NAT-related media drops and ensures predictable firewall rules

8. Support Channels

Provisioning Support:

Contact your Exotel CSM or email: [email protected]

Technical Support:

Visit: support.exotel.com

Share the following:

Account SID

Trunk transport type (TCP/TLS/FQDN)

Sample Call SIDs

SIP trace logs (from sngrep/Wireshark)