INFO

FAQ SwyxWare Version 5.0 (kb2980)

The information in this article applies to:

  • SwyxWare v5.00

[ Summary | Information ]


Summary

This article gives you a brief overview of all new features of SwyxWare version 5.0 and answers frequently asked question.


Information

Which new features does SwyxWare version 5.0 offer?

 

The significant feature of SwyxWare version 5.0 is the SIP trunking functionality (see below). Other new features are:

 

  • Support of ENUM (see below)
  • Compression G.729a (see below)
  • Forwarding of calls before pickup
  • Import and export function for the personal phonebook
  • HiFi ringtones
  • Support of new devices (SwyxPhone L520s/L540, extension module T520, wireless headset H360)
  • Skin with caller protocol
  • Controller for easy adjustment of the volume

 

 

What is SIP trunking?

 

Trunking refers to the connection between the PBX and a service provider, e.g. an ITSP (Internet Telephony Service Provider) or SIP provider. Calls are processed with the communication protocol SIP (Session Initiation Protocol). In SwyxWare the connection is configured as a SwyxLink.

 

Which SIP providers can be recommended?

 

When looking at providers of internet telephony, it is important to distinguish between offers for private and business customers. Most providers that are today focussed on private customers are starting to offer solutions for business customers as well. This also applies for the hosting of DDI (Direct Dial In) support.

Providers of internet telephony are for example:

  • Austria: Inode.at (supports DDI)
  • Germany: web.de, purtel.de, nikotel.de, Sipgate, GMX, 1&1, freenet
  • UK: Calluk, Pipemedia
  • US: Vonage 
  • Information about SIP provider in other countries will follow.

 

 

How can I save telephone costs with the connection to a SIP provider?

 

In first place costs can be saved by using true internet telephony, meaning both callers are using the internet, e.g. the same service provider. This way there are no additional phone costs charged.

When calling somebody in the conventional telephone network over a service provider, costs are the same as with a traditional telephone provider. Most of the time the costs per minute are lower, but the same savings can also be made by using call-by-call and pre-selection services.

 

 

Can I send faxes over a SIP connection?

 

Yes you can, if your provider offers the standard T.38, like e.g. SwyxWare does. Unfortunately we don’t know about a provider which offers T.38 for secure fax transmission at this time.

 

 

Do I still need my ISDN line?

 

In the long run all phone calls will be handled over IP. Today there are already companies that provide business customers with an “IP-only” offer. But the realisation of a connection with full DDI support similar to the ISDN PRI, is still in the planning stage for a lot of providers. Therefore the connection to a SIP service provider has to be seen as an additional connection type to PSTN to save communication costs of companies.

 

 

What is ENUM?

 

ENUM (tElephone NUmber Mapping) is a specification to translate telephone numbers to domain names. ENUM is the Internet Engineering Task Force (IETF) protocol that will assist in the convergence of the PSTN and the IP network. It is the mapping of a telephone number from the PSTN to Internet services: telephone number in, URL out. ENUM was developed as a solution to the question of how to contact a user over the same number in the internet and the PSTN.

A call to a phone number that has an ENUM domain can be either forwarded to the PSTN, a mobile phone or an IP number. Any phone number, such as +1 555 42 42 can be transformed into a hostname by reversing the numbers, separating them with dots and adding the e164.arpa suffix, like so: 2.4.2.4.5.5.5.1.e164.arpa

 

 

How much bandwidth do I have to reserve for the SIP link and what codec is being used?

 

It depends on the service provider whether the files are being compressed or not. In the terms of use of the service provider you can also find out which codec has to be used. Codec G.711 requires 80kBit/s and codec G.729a between 18 and 40kBit/s. For SIP service providers the codecs G.711 A-law, G.711 µ-law and G.729a are supported at the moment.

 

Hint: With the VoIP bandwidth calculator you can calculate the bandwidth you require (http://www.packetizer.com/voip/diagnostics/bandcalc.html).

 

What bandwidth does codec G.729a require?

 

The net value G.729a requires is app. 8 kBit/s, but the gross value is heavily dependent on the packet size which differs with every provider.

A standard packet size of 20 ms – which is used in SwyxWare – adds up to a gross demand of 24 kBit/s (30 ms to 18 kBit/s).

 

 

Which device determines when and which compression is being used?

The service provider and SwyxWare decide on the compression. In the SwyxWare administration you can define which codec should be used between SwyxServer and the service provider. Otherwise the general setting is codec G.711 which gives you the best voice quality.


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