Routes: End-Point

AG Projects RTC Platforms (Routing Guides)

This guide describes the routing from and to an End-Point.

End-Point to End-Point

This document describes the routing for End-Point to End-Point

Quick Dial

Quick dial is a per SIP account feature that allows to dial short numbers to match other SIP accounts in the same number range. The SIP Proxy will try to autocomplete the number to form a full address. To use this feature:

1. The username part of the SIP account must be numeric (example 31208005169@ag-projects.com)
2. The quickdial attribute of the SIP account must be set to a substring matching the beginning of the username (e.g. 312080051).
3. When user dials 60 the example above, the SIP Proxy will concatenate the quickdial set to 312080051 with the dialed number 60 and try 31208005160@ag-projects.com as destination.

End-Point to PBX

End-Point to PSTN

For interconnection with PSTN, a SIP trunking service must be setup between the SIP Proxy and the PSTN gateway provider. The authorization of SIP requests is based on transitive trust. The SIP Proxy has a trust relationship with each SIP subscriber and the PSTN gateway has a trust relation with the SIP Proxy. The trust relation between the SIP Proxy and the PSTN gateway is based on the IP addresses. The PSTN gateway cannot use DIGEST authentication in the relation with the SIP Proxy because it does not have access to the SIP accounts database of the SIP Proxy.

PSTN Gateway Requirements

Must have:

• SIP signaling based on RFC 3261
• DNS lookups based on RFC 3263
• Support for SIP extensions for caller id and privacy (P headers)
• RTP active mode (send RTP data as soon as call setup is completed)
• Use public routable IP addresses for both signaling and media

Recommended:

• ENUM lookups based on RFC 3761

Routing to PSTN destinations is realized by provisioning the PSTN carriers, gateways and routes (also known as Least Cost Routing engine or LCR). The structure of the PSTN provisioning is as follows:

Route → Carriers → Gateways → Rules

For each PSTN prefix (called a PSTN route) a set of carriers can be assigned with an optional priority. Each carrier can have one or more gateways and each gateway can have optional rules for converting the number. For more information see the provisioning guides.

Once the SIP request is authenticated, the SIP Proxy authorizes the request based on the rights associated with the subscriber account and decides whether a SIP session to the PSTN gateway is allowed or not. If the session is allowed, the SIP Proxy asserts an identity associated to the SIP account, which can be the telephone number presented as caller ID to the destination, locates a PSTN gateway for the dialed number (by using least cost routing or other configured logic) and forwards the request to the PSTN gateways inserting itself in the path of subsequent messages.

Caller id indication

The platform generates a Caller ID indication by appending Remote-Party-Id or P-Asserted identity headers, depending on its configuration. The content of the headers is generated from:

SipAccount → rpid attribute associated with the SIP account

Next Steps

Continue by:

• learning about Routes: MSRP media or
• learning about Routes: PBX or Trunk or
• learning about Routes: MS Teams; or
• learning about Routes: Presence or
• learning about Routes: PSTN or
• learning about Routes: Special or
• going back to the Overview.