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Note that changing from FXO to FXS is not something you can simply do with a set‐ |
tings change. FXO and FXS ports require completely different electronics. |
This stuff is old-school, folks. You can run old phones from 100 years ago off an FXS |
port! |
We do not recommend the use of analog trunks in an Asterisk system. Their configu‐ |
ration and use is outside of the scope of this book.4 |
4 We should note, however, that we’ve written extensively on the subject in the past, and that body of work has |
been released under Creative Commons licensing, and is freely available. |
112 |
| |
Chapter 7: Outside Connectivity |
Digital telephony |
Digital telephony was developed in order to overcome many of the limitations of |
analog. Some of the benefits of digital circuits include: |
• No loss of amplitude over long distances |
• Reduced noise on circuits (especially long-distance circuits) |
• Ability to carry more than one call per physical circuit |
• Faster call setup and teardown |
• Richer signaling information (especially if using ISDN) |
• Lower cost for carriers |
• Lower cost for customers (at higher densities) |
There were several fundamental digital circuits that gained wide acceptance in the |
telecommunications industry: |
T1 (24 channels) |
Used in Canada and the United States (mostly for ISDN-PRI)5 |
E1 (32 channels) |
Used in the rest of the world (ISDN-PRI or MFC/R2) |
BRI (2 channels) |
Used for ISDN-BRI circuits (Euro-ISDN) |
Note that the physical circuit can be further defined by the protocol running on the |
circuit. For example, a T1 could be used for either ISDN-PRI or CAS, and an E1 |
could be used for ISDN-PRI, CAS, or MFC/R2. |
It is difficult to justify these circuit types anymore. Relative to VoIP protocols, they |
have become expensive, complex, and somewhat inflexible. If you need to connect |
such circuits to an Asterisk system, we recommend some sort of gateway device to |
convert the circuit into SIP, and then connect via SIP to your Asterisk system. If you |
want a single-chassis system, companies such as Digium and Sangoma offer digital |
PSTN cards that can be installed directly into your Asterisk server; they are |
5 There’s also a circuit used in Japan called a J-1, which is most simply described as a 24-channel E1. |
The PSTN |
| |
113 |
connected to Asterisk by way of the DAHDI channel driver. The use of this technol‐ |
ogy is outside of the scope of this book.6 |
VoIP |
Compared to the lengthy history of the telecommunications industry,7 VoIP is still a |
relatively new concept. For the century or so prior to VoIP, the only way to connect |
your site to the PSTN was through the use of circuits provided for that purpose by |
your local telephone company. VoIP now allows for connections between endpoints |
without the PSTN having to be involved at all (although in most VoIP scenarios, |
there will still be a PSTN component at some point, especially if there is a traditional |
E.164 phone number involved). The PSTN still controls the phone numbers, and we’ll |
be using those until somebody comes up with an internet-based addressing mecha‐ |
nism that is not subject to abuse the way email has been.8 |
Network Address Translation |
If you are going to be using VoIP across any sort of wide-area network (such as the |
internet), you will be dealing with firewalls, and quite frequently network address |
translation (NAT) as well.9 A basic understanding of how the SIP and RTP protocols |
work together to create a VoIP call can be helpful in understanding and debugging |
functional problems (such as the “one-way audio” issue NAT configuration errors can |
often create). NAT allows a single external IP address to be shared by multiple devices |
behind a router. Since NAT is typically handled in the firewall, it also forms part of |
the security layer between a private network and the internet. |
A VoIP call using SIP doesn’t consist just of the signaling messages to set up the call |
(the SIP part of the connection). It also requires the RTP streams (the media), which |
carry the actual audio connection,10 as shown in Figure 7-1. |
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