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
3. Transport Options
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)
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 hello@exotel.com (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
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