Home / Solutions / EnableX – SIP/PSTN Integration Solution

In addition to calling via advanced browsers and mobile apps, EnableX provides a way to connect the conference from mobile phones, PSTN (a.k.a Landline) phones or Enterprise SIP infrastructure through its SIP interface. EnableX is standard SIP RFC 3261 compliant and can connect to SIP phone or Enterprise SIP infrastructure. EnableX supports Inbound and Outbound dialling for SIP/PSTN side.

At this point, EnableX supports Audio Only calls and does not support Video.

EnableX hosts https://sip.enableX.io cluster, that can be expanded as per the requirements. https://sip.enablex.io hosts multiple EnableX SIP servers and transcoders (Currently only between Opus and G711) that are connected to EnableX WebRTC platform.

VCloudX, the company of EnableX, has partnered with Telephony Service Provider to terminate the calls on to PSTN or GSM mobile networks, needless to mention that these devices can call inbound to EnableX conferences

Architecture Diagram

General requirements to enable SIP calling on EnableX platform

  • Enabling SIP room: While creating room, developer passes sip enabled flag in JSON Payload as { sip: { enabled: true }}.
  • EnableX supports RFC-2833 for the DTMF detection
  • If you are an enterprise customer with existing SIP infrastructure, you need to contact EnableX support to get your SIP outbound proxy to be whitelisted.
  • Incoming SIP calls to EnableX must have an indication of the caller id, either in From header or one of the following headers P-Preferred-Identity:, P-Asserted-Identity:, Remote-Party-Id
  • Outgoing SIP/PSTN call identity will be dependent on the service provider that is available in the region.
  • You will be able to submit your own IVR!
  • For PSTN dial-in, please contact support@enablex.io to receive the dial-in number.
  • There is no DID facility to dial directly into the conference

Workflow

Incoming

The success of the room creation will return SIP PIN, which user needs to enter when dialling into the conference.

Once the Room is created, EnableX automatically creates a route between EnableX conference and EnableX SIP gateway.

The Enablex PBX(Asterisk) is set up with a dial-in number for each partner who has opted for SIP Service. This IVR (@Asterisk PBX) can be configured to have more personalized IVR messages for the partner.

There is NO DID for the conference, so it is mandatory to provide the Conference Id (provided as PIN) once you are connected to the platform.

Enterprise customers with existing SIP Infrastructure

For the partner SIP infrastructure to be integrated with EnableX SIP Server, the partner must provide the external IP of the SIP infrastructure which will be used for authenticating any calls coming into our EnableX server (Whitelist).

Dialling Information

For a partner with existing SIP Infrastructure, the SIP extension to dial will be configured during the initial onboarding of the partner. The user will be asked for additional configuration of IVR for personalized greeting, which can be uploaded through the portal.

When a SIP room is created using the APIs, the user will be sent with the following details for joining the SIP Room.

Dialling info: Extension to dial with externally reachable sip details of the EnableX.

SIP PIN: This is created dynamically during the room creation, and each user has to enter the pin via DTMF to connect to EnableX rooms.

Dialling in From PSTN

Partner need to opt for a SIP Service with PSTN connectivity to enable this service. This service can be enabled with features such as how many dial-in numbers partner requires. These dial in number can be provisioned to be associated with multiple application if the user has more than one application which needs SIP service the portal needs to be modified to support that The trunk number to dial is enabled. Partner IVR and extension to dial in from SIP is configured in EnableX PBX (asterisk) as detailed out the in the previous section.

When a SIP room is created using the APIs, the user will be sent with the following details for joining the SIP Room.

Dialing info: PSTN (trunk) number to dial SIP PIN.

SIP PIN: This is created dynamically during the room creation, and each user has to enter the pin via DTMF to connect to EnableX rooms.

Deployment

EnableX thrives to provide the easy integration/adaptation to the external world. EnableX hosts an Asterisk complaint PBX with an IVR. As a service provider, or as an enterprise, if you have PBX already setup, you can easily create a SIP Trunk between these two entities.