Documentation for developers

SIP

Introduction

Welcome to the Speaqr SIP phone service. Our SIP server allows you to make phone calls with real-time translation, breaking language barriers in phone communications.The service uses SIP over TLS technology to ensure secure communications, along with SRTP for audio encryption and the μ-law (PCMU) codec for universal compatibility with telephony systems.With our SIP service, you can integrate Speaqr's translation capabilities directly into your PBX, softphone, or any SIP-compatible system, allowing multilingual calls without the need for physical interpreters.To use the SIP service you need valid access credentials. To obtain them, contact us through the contracting form.

Prerequisites

Before setting up your SIP client, make sure you meet the following requirements:

Compatible SIP client

Your SIP client must support SIP over TLS for secure signaling, SRTP (SDES) for audio encryption, and the μ-law (PCMU) codec, which is mandatory for service compatibility.Examples of compatible clients: Any softphone or PBX with support for PJSIP.

Access credentials

You will receive a username (e.g. new-user), a password, and the server domain: sip.speaqr.ai.

Network connection

Make sure that the ports 5061/TCP (for TLS) and 10000-20000/UDP (for SRTP) are open on your network or firewall.

TLS Certificates

You don't need to generate additional certificates. Our server uses a valid TLS certificate. Make sure your client accepts standard trusted certificates.

Customer configuration

Set up your softphone or PBX to connect to sip.speaqr.ai with the following parameters:

Account information

Username: new-user (provided by the administrator)
Password: Your provided password
Domain/Server: sip.speaqr.ai:5061

Protocol configuration

Transport: TLS
Port: 5061
Media Encryption: SRTP (SDES)
Codecs: Enable only PCMU (μ-law); disable other codecs

Network options

Register on the server: Activate the register option (REGISTER)
STUN (if you are behind NAT): Use a STUN server like stun.l.google.com:19302

Custom SIP headers

Set up the ability to send custom SIP headers (check your client's documentation to add them).Example: X-Origin: new-user, X-Destination: destination

Services available

The server offers two services through the ARI loopback application. Use the following methods to interact with them:

Loopback service

Send audio to the server and receive the audio processed (intervened) by Speaqr.

How to use it

Mark the loopback extension in your SIP client.

Required SIP headers

X-Origin: Your identifier (e.g. new-user)
X-Destination: An identifier for the destination (can be any relevant value, e.g. test)

Behavior

The server responds with the audio processed by the Speaqr application, based on your email (obtained from the database) and the headers.

Call Service

Make a call to a physical phone number, with the audio intervened by Speaqr.

How to use it

Dial the call extension on your SIP client.

Required SIP headers

X-Origin: Your identifier (e.g. new-user)
X-Destination: The physical phone number you wish to call (international format, e.g. +1234567890)
X-Dial: Optional, to specify an additional number or destination (e.g. +1234567890)

Behavior

The call connects to the specified number, with the audio processed by Speaqr.

Troubleshooting

If you encounter difficulties during the setup or use of the SIP service, check the following solutions:

The account is not registered

• Verify that the username, password, and port (5061/TLS) are correct
• Ensure that the firewall allows port 5061/TCP
• If you are behind NAT, enable STUN or contact the administrator

No audio

• Confirm that SRTP is enabled and that the RTP ports (10000-20000/UDP) are open
• Make sure that only PCMU is enabled as a codec

Error using loopback or call services

• Verify that the X-Origin and X-Destination headers are correctly configured
• Make sure your account is registered before dialing
• Contact the administrator if the ARI application does not respond

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