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

1. What is Virtual SIP Trunking (vSIP)?

Virtual 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 200 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, GJ, WB

Getting Started with Exotel vSIP (Veeno Accounts)

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

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

  3. Procure Exophones by emailing hello@exotel.com (mention region-specific need)

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

  5. 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

Share Configuration Details:

  • 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 Virtual 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 Virtual 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 Virtual 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: hello@exotel.com

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)


Version: Master-vSIP-Guide
Last Updated: June 2025