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SIP & PSTN

This API supports dialing through a simple phone number - PSTN or a Voice Over IP system - SIP endpoint. If you don't have your own Voice over IP (VoIP) system, you will want to use a phone number to make the connection.

SIP (Session Initiation Protocol)#

Session Initiation Protocol (SIP) is a standardized communications protocol that has been widely adopted for managing multimedia communication sessions for voice and video calls. SIP may be used to establish connectivity between your communications infrastructures and Symbl's communications platform.

info

SIP captures audio quality at 16KHz and above.

{
"endpoint": {
"type": "sip",
"uri": "sip:555@your_sip_domain", // SIP URI to dial in
"audioConfig": { // Optionally any audio configuration
"sampleRate": 16000,
"encoding": "PCMU",
"sampleSize": "16"
}
}
}

PSTN (Public Switched Telephone Networks)#

The Publicly Switched Telephone Network (PSTN) is the network that carries your calls when you dial in from a landline or cell phone. It refers to the worldwide network of voice-carrying telephone infrastructure, including privately-owned and government-owned infrastructure.

info

PSTN captures audio quality at 8KHz max.

{
"endpoint": {
"type": "pstn",
"phoneNumber": "PHONE_NUMBER", // Use international code.
"dtmf": "DTMF_MEETING_ID" // if password protected, use "dtmf": "<meeting_id>#,#<password>#"
}
}
  • SIP (Session Initiation Protocol)
  • PSTN (Public Switched Telephone Networks)