Calling from PSTN to VoIP

As VoIP services are growing, traditional (PSTN) and IP telephony need to co-exist. Users can belong either to one or the other network, and inter-working between the two technologies is necessary. That requires translation between the different protocols used, which is provided by signaling/media gateways. This post makes first a quick introduction to the signaling process of a PSTN call, and then it describes a call scenario where a PSTN subscriber calls a VoIP user.

Traditional telephony (PSTN network) uses a signaling protocol called ISDN User Part (ISUP). When a user initiates a call, an Initial Address Message (IAM) is sent in order to reserve a circuit for the call. The destination switch receives the IAM, initiates ringing to the called party and sends back to the caller an Address Complete Message (ACM). When the called party picks up the phone, an Answer Message (ANM) is sent back to the caller. Finally, when a user terminates the call, a Release (REL) message is sent to the other party, who replies back with a Release Complete (RLC) message.

In the VoIP environment, the SIP protocol is used as a signaling protocol. A basic call flow of this signaling process is described here.

Now when a PSTN subscriber calls a VoIP user, special signaling and media gateways perform the translation and mapping between ISUP and SIP messages. The picture below shows a basic ISUP & SIP call flow:

VoIP-PSTN Interworking. Translation between ISUP and SIP.

In this simple call scenario, you can see the direct relation between the different signaling messages used for initiation, answer and termination of the call. Since ISUP and SIP protocols consist of various responses and error/release codes, different mapping techniques exist to properly translate those messages.

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About TelcoNotes

IP & VoIP networking

Posted on September 1, 2013, in VoIP and tagged , , , , . Bookmark the permalink. 7 Comments.

  1. Hi,
    Calling from PSTN and VoIP and vice-versa is possible. Can you post an article on how the communication is possible from packet switching and circuit switching in a simple way?

  2. Please post a complet Asterisk help

  3. Is the method shown here is for SIP-I or SIP-T??

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