This guide outlines how to integrate Exotel’s Virtual SIP Trunking (vSIP) with your voice application flow (e.g., IVR or agent connect) using SIP signaling. It focuses on the use case of routing inbound SIP calls from external SIP systems into Exotel Flow Applets

1. Use Case: External SIP System to Exotel Flow

This guide is meant for customers who:

  • Host their own SIP PBX or Bot Platform.

  • Want to initiate SIP calls into Exotel to trigger flows (like Connect or IVR).

  • Require SIP-over-TLS or TCP transport.

Example Scenarios

Scenario

Description

SIP Bot to Exotel Flow

External SIP bot initiates call to Exotel VN → Exotel triggers IVR flow

SIP PBX to Exotel Flow

SIP endpoint dials into Exotel Flow for agent routing or IVR processing

Regional SIP→Flow (cross-border PoP)

Enterprise infra routes call to Exotel PoP (SG/Mum/KA) for flow handling

2. SIP Integration Requirements

What You Need to Share with Exotel

  • Account SID

  • Exophone (VN) that should route to flow

  • Source IP or FQDN from where SIP INVITE will originate

  • Transport type (TCP/TLS)

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 Exotel Will Do

  • Provision SIP trunk mapping on the VN

  • Route incoming INVITE on SIP trunk to your assigned Flow (via Connect Applet)

3. SIP Configuration Reference (Customer Side)

Here is an Asterisk-style example for TLS:

[general]

externip = <your_public_ip>

localnet = 192.168.0.0/16


[exotelvsip]

type = friend

context = outgoing

fromdomain = <accountsid>.pstn.exotel.com

host = pstn.sgp1.exotel.com

port = 443

transport = tls

disallow = all

allow = ulaw

allow = alaw

nat = force_rport

insecure = port

canreinvite = no

sendrpid = yes

trustrpid = yes

relaxdtmf = yes

encryption = yes

Ports to Open

Type

Port

Protocol

Description

SIP

443 / 5070

TLS/TCP

SIP signaling

RTP Media

10000-40000

UDP

Media/audio ports

4. Sample SIP INVITE to Exotel (Flow Integration)


Sip                                        CopyEdit

INVITE sip:+91XXXXXXXXXX@pstn.sgp1.exotel.com SIP/2.0

Via: SIP/2.0/TLS your-sbc.domain.com:5061;branch=z9hG4bK4041f853

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

To: <sip:+91XXXXXXXXXX@pstn.sgp1.exotel.com>

Call-ID: abcdef1234567890@exotelt.pstn.exotel.com

CSeq: 102 INVITE

Contact: sip:+91YYYYYYYYYY@your-sbc.domain.com:5061;transport=tls

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

Supported: replaces, timer

X-Exotel-AccountSid: exotelt

Content-Type: application/sdp


Header Reference – SIP INVITE to Exotel (Flow Integration)

Header

Mandatory

Example (Redacted)

Description

Request URI

Yes

sip:+91XXXXXXXXXX@pstn.in2.exotel.com

Target Exophone (VN) to route the call into a Flow or PSTN leg.

X-Exotel-AccountSid

Yes

Exxxxxx

Identifies your Exotel account. Required for Flow integration.(Tenant name)

Via

Yes

SIP/2.0/TLS your-sbc.domain.com:5061

Your SBC’s transport, used by Exotel to route responses.

From

Yes

"Sales" <sip:+91YYYYYYYYYY@customer.com>

CLI of Customer(Callee)

To

Yes

<sip:+91XXXXXXXXXX@pstn.in2.exotel.com>

Exophone

Contact

Yes

<sip:+91YYYYYYYYYY@your-sbc.domain.com:5061>

Your contact URI for Exotel to send 200 OK, BYE, and other dialog messages.

Call-ID

Yes

abc123xyz@your-sbc

Unique identifier for SIP session; echoed in Exotel response.

CSeq

Yes

102 INVITE

SIP sequence number.

Max-Forwards

Yes

70

Prevents SIP loops by limiting hops.

Supported

Optional

replaces, timer

Advertises SIP extensions your UA supports.

Allow

Optional

INVITE, ACK, BYE, CANCEL, OPTIONS

SIP methods your SBC is willing to process.

Content-Type

Yes

application/sdp

Indicates the message contains SDP for media negotiation.

Important Notes

  • Only use X-Exotel-AccountSid for Flow routing. This is essential for Exotel to associate the call with the correct Flow.

  • Do not add X-Exotel-CallSid, X-Exotel-LegSid, or X-Exotel-TrunkSid. These headers are auto-injected by Exotel in the inbound leg.

  • Ensure your SIP infrastructure handles:

    • G.711 PCMA (preferred codec)

    • SIP over TLS (port 443)

    • SRTP if TLS is enabled

5. Edge PoPs and IPs to Whitelist

Region

SIP Proxy FQDN

Media IPs

Karnataka

pstn.in1.exotel.com

14.194.10.247, 61.246.82.75

Mumbai

pstn.in2.exotel.com

182.76.143.61, 122.15.8.184

Singapore

pstn.sgp1.exotel.com

18.142.150.245, 122.248.223.73

KSA (upcoming)

coming soon

coming soon

6. Testing Steps

  • Place a SIP INVITE to the Exophone you've been assigned.

  • Confirm that the call lands in the expected flow (e.g., Connect applet rings agent).

  • Use sngrep, sip set debug on, or Wireshark to validate SIP path.

  • Ensure RTP is established from your system to Exotel's media IP.

7. Best Practices

  • Ensure Exophone is mapped to a live flow before testing.

  • Always share your SIP signaling IP/FQDN and transport in advance.

  • Prefer PCMA (G.711 A-law) for better regional PSTN compatibility.

  • Use TLS transport when possible to ensure encryption and SRTP.

  • Confirm DNS resolves FQDN correctly if not using static IP.

8. Support

  • Contact your Exotel Account Manager with:

    • Account SID

    • Flow/Exophone ID

    • SIP signaling IP or FQDN

    • Preferred Transport (TCP or TLS)

  • Technical Help:

    • support.exotel.com

    • Share call samples and SIP logs for troubleshooting

Version: vSIP-to-Flow-Guide (Alpha)
Last Updated: June 2025