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 tosip.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 theloopback
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 thecall
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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