This article provides enterprise customers with technical steps to integrate Exotel’s Virtual SIP Trunking (vSIP) in Alpha with your PBX, SIP server, or SBC over TCP (Mumbai PoP). It outlines configuration practices, header formats, best practices, and validation techniques.

1. Product Overview

Exotel’s Virtual SIP Trunking (Alpha) allows secure SIP-based PSTN call origination and termination between your SIP infrastructure and Exotel’s platform using IP authentication. This Alpha release is intended for controlled pilots and is not SLA-backed for production workloads.

2. Architecture

  • Call Type: PSTN <-> SIP Gateway Interconnect

  • Transport: SIP over TCP (Port 5070)

  • Media: RTP over UDP (Ports 10000–40000)

  • Authentication: IP Whitelisting (Registration-based auth not supported)

  • Edge Location: Mumbai PoP (India)

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.

3. Required Configuration

IP Whitelisting

  • Your static public IP must be whitelisted by Exotel.

  • Registration (SIP REGISTER) is not supported.

Ports to Open

Type

Port Range

Protocol

Purpose

Signaling

5070

TCP

SIP signaling

Media

10000–40000

UDP

RTP streams

SIP Domain and Proxy Details

Media Server POP

Region

Media IPs

Mumbai DC

182.76.143.61, 122.15.8.184

KA DC

14.194.10.247, 61.246.82.75


Signaling Server POP


Region

Proxy FQDN

Mumbai DC

pstn.in2.exotel.com

Mumbai Cloud

pstn.in4.exotel.com


Use this proxy for configuring Exotel as a peer/trunk on your SBC or PBX.

4. Sample Configuration – Asterisk PBX

[general]

externip = <your_public_ip>

localnet = 192.168.0.0/16


[exotelvsip]

type = friend

context = incoming

fromdomain = <accountsid>.pstn.exotel.com

host = pstn.in2.exotel.com

port = 5070

transport = tcp

disallow = all

allow = alaw

allow = ulaw

nat = force_rport

insecure = port

canreinvite = no

sendrpid = yes

trustrpid = yes

relaxdtmf = yes

transport=tcp


5: SIP Message Format

A. INVITE from Exotel Trunk (Exotel → Customer)

This is triggered when Exotel routes an inbound call to the customer SIP gateway over TCP.

Sample SIP INVITE

Sip                                            CopyEdit

INVITE sip:+91XXXXXXXXXX@<customer-ip>:5061;transport=tcp SIP/2.0

Record-Route: sip:<exotel-ip>:443;transport=tls;lr

Via: SIP/2.0/TCP <exotel-ip>:443;branch=z9hG4bK2414...

From: "+91AAAAAAAAAA" <sip:+91AAAAAAAAAA@exotelt.pstn.exotel.com>;tag=as2aefddf2

To: <sip:+91XXXXXXXXXX@<customer-ip>>

Call-ID: <UUID>@pstn.mum1.exotel.com

CSeq: 102 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces

X-Exotel-LegSid: <leg-id>

X-Exotel-CallSid: <call-id>

X-Exotel-TrunkSid: <trunk-id>

P-Asserted-Identity: <sip:+91AAAAAAAAAA@exotelt.pstn.exotel.com>

P-Early-Media: supported

Contact: <sip:+91AAAAAAAAAA@<public-ip>:port;transport=tcp>

Content-Type: application/sdp

Content-Length: 1168

Max-Forwards: 67


v=0

o=root 1683048786 1683048786 IN IP4 <exotel-media-ip>

c=IN IP4 <exotel-media-ip>

t=0 0

m=audio 37456 RTP/AVP 8 0 96

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

a=sendrecv

a=rtcp:37457

a=ptime:20


Header Reference Table – INVITE from Exotel

Header

Mandatory

Description

Request URI

Yes

Destination SIP URI (Exophone mapped to customer SIP IP).

Via

Yes

Routing path for SIP responses. Uses TCP in this case.

Record-Route

Optional

Ensures Exotel remains in signaling path for subsequent requests.

From

Yes

Caller party number (CLI) received by Exotel (e.g., end-user).

To

Yes

Exophone provisioned by Exotel for this customer.

Call-ID

Yes

Unique identifier for the call.

CSeq

Yes

Command sequence number, must increment with each new transaction.

Allow

Yes

List of SIP methods supported by Exotel.

Supported

Optional

SIP extensions like replaces, timer.

X-Exotel-CallSid

Yes

Unique ID for the call session (tracking/debugging).

X-Exotel-LegSid

Optional

Call leg identifier; unique to this direction of the call.

X-Exotel-TrunkSid

Optional

Identifies which virtual SIP trunk was used.

P-Asserted-Identity

Optional

Validated caller ID presented to customer SIP server.

P-Early-Media

Optional

Indicates support for early media before call is answered.

Contact

Optional

Where Exotel can be reached for further in-dialog SIP messages.

Content-Type

Yes

Indicates media description follows (SDP).

Content-Length

Yes

Length of the SDP body.

Max-Forwards

Yes

Prevents infinite loops by limiting hop count.

SDP (v=...)

Yes

Contains RTP setup: IP, codecs (PCMA/PCMU), RTP/DTMF, etc. (unencrypted RTP).

B. INVITE to Exotel Trunk (Customer → Exotel)

Customer initiates an outbound call over TCP, using a registered Exophone as the CLI and targeting any mobile/landline via Exotel.

Sample SIP INVITE

Sip                                                                                  CopyEdit

INVITE sip:+91YYYYYYYYYY@<exotel-ip>:5070 SIP/2.0

Via: SIP/2.0/TCP <customer-ip>:5061;branch=z9hG4bKbK4041f853

Max-Forwards: 70

From: "+91XXXXXXXXXX" <sip:+91XXXXXXXXXX@exotelt.pstn.exotel.com>;tag=as63e4d7f1

To: <sip:+91YYYYYYYYYY@<exotel-ip>>

Contact: <sip:+91XXXXXXXXXX@<customer-ip>:5061;transport=tcp>

Call-ID: <UUID>@exotelt.pstn.exotel.com

CSeq: 102 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 371


v=0

o=root 1002281923 1002281923 IN IP4 <customer-media-ip>

c=IN IP4 <customer-media-ip>

t=0 0

m=audio 18232 RTP/AVP 8 0 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=maxptime:150

a=sendrecv


Header Reference Table – INVITE to Exotel

Header

Mandatory

Description

Request URI

Yes

Final destination number (callee) to be dialed via Exotel trunk.

Via

Yes

Describes how to return SIP responses to customer (over TCP).

From

Yes

CLI being used — must be a valid Exophone registered with Exotel.

To

Yes

Destination number (callee, Called Party Number); not used for routing.

Call-ID

Yes

Unique identifier for the call session.

CSeq

Yes

SIP sequence number and method (e.g., INVITE).

Contact

Optional

Where Exotel can reach back for in-dialog requests.

Allow

Yes

Methods supported by customer.

Supported

Optional

SIP extensions.

Content-Type

Yes

MIME type of message body (SDP).

Content-Length

Yes

Byte length of SDP content.

Max-Forwards

Yes

Used to control and limit SIP hops.

SDP (v=...)

Yes

Media negotiation block, including codecs and ports. RTP only (unencrypted).

Recap – CLI & Exophone Roles

Direction

From (CLI)

Request URI (Destination)

Notes

Exotel → You

Customer’s end-user CLI

Your SIP URI (Exophone)

Used in inbound flows to your system

You → Exotel

Your Exophone (CLI)

Final user number to dial

Must use registered CLI in From, else call will be rejected


6. Best Practices and Pre-checks

Before Configuration

  • Confirm you have a static public IP with no CG-NAT or port masking.

  • Make sure your firewall allows TCP 5070 and UDP 10000–40000 bidirectionally.

  • Validate that your PBX/SBC supports SIP over TCP and G.711 codecs (PCMA/PCMU).

During Configuration

  • Prefer PCMA as the first codec in media negotiations; PCMU as fallback.

  • Disable registration and use static peer trunking.

  • Use nat=force_rport for NAT traversal if behind a firewall.

  • Monitor and log Call-ID, X-Exotel-* headers for debugging.

After Configuration

  • Run test calls and inspect SIP INVITEs and RTP.

  • Monitor audio quality, latency, and early media (ringback).

  • Use tcpdump or sngrep to validate SIP dialogues and media ports.

Quick sanity checks

  1. Transport match – your Via: …;transport=tcp must align with port 5070.

  2. Caller-ID hygiene – the From: number in your outbound INVITE must be an authorised CLI/Exophone.

  3. Log Exotel headers – X-Exotel-CallSid, LegSid, TrunkSid are read-only but invaluable for troubleshooting.


7. How to Test Your Setup

Inbound Test (Exotel → Your SIP Server)

  • Trigger a call to your Exotel VN mapped to the SIP trunk.

  • Check if your server receives an INVITE from pstn.in2.exotel.com.

  • Confirm RTP is flowing from 182.76.143.61 or 122.15.8.184.

Outbound Test (Your Server → Exotel)

  • Send a SIP INVITE to pstn.in2.exotel.com:5070 with the To URI set to a valid destination (e.g., mobile number).

  • Ensure correct formatting of headers and the contact field.

  • Look for 100/180/200 OK responses and RTP flow.

8. Troubleshooting Tips

Issue

Potential Cause

Resolution

No INVITE received

IP not whitelisted

Contact Exotel to confirm IP ACL

Call drops in 30s

NAT binding lost

Enable symmetric RTP / force_rport

One-way audio

Media ports blocked

Open UDP 10000–40000

403 Forbidden

Incorrect domain or auth

Use correct fromdomain and no auth creds

9. Support and Next Steps

This guide covers Exotel vSIP over TCP via the Mumbai PoP under the Alpha release. Production deployment readiness (with TLS, SRTP, failover, dashboards) will be announced during the GA phase.

For support:

  • Contact your Exotel account manager.

  • Or raise a ticket via https://support.exotel.com with:

    • Account SID

    • Call time and number

    • SIP trace logs (.pcap or .txt)

Version: Alpha-TCP-Mumbai | Last updated: June 2025 Exotel reserves the right to modify Alpha features and configurations without notice.