 Should I use a high-speed connection or dial-up connection?
We recommend that you use VoIP services only if you have a high speed
Broadband Internet connection, as the quality of the calls depends mostly on the Internet your connection speed. Tests conducted on VoIP using dial up services have produced dropped calls, call lagging, and other call quality issues.
What do I need to make Net4 VoIP work?
There are several configurations for VoIP, and all require you to have an account with Net4.
From PC: If you wish to make calls from your PC only, then you can download software to do this from the following link: www.phonewala.com
Add a headset or a microphone and speakers, and you are ready to call any number in the world.
Phone-to-Phone: You can also get an Analogue Telephone Adapter (ATA) that connects to high speed modem and then via the Internet to the PSTN network.
What advanced features does VoIP offer?
Apart from saving money, the best reason for using VoIP is the host of new services that come with it. These include:
- Multiple Area Codes. Add multiple inbound area codes for your phone, allowing you to create a 'virtual presence' in any part of the world, or provide a local calling number to family members.
- Conference call. Have a conference call with multiple people on the same call.
- Individual Call Filtering and Blocking. Filter and block unwanted calls using this Caller ID-like facility.
- Call Hunting/Find Me. Allow specified individual numbers specify to find you wherever you are (either one at a time in sequence, or all at the same time).
- Voicemail. Receive voicemail messages in your e-mail inbox or on your web page.
- Mobility. If you use an ATA, you can take it with you when you travel and use it wherever there is a high speed Internet connection.
How do I make a phone call?
Pick up your phone and dial 1 + area code + phone number. You must dial 1 and then the area code for all calls, even for the local calls.
Can I make calls if my Internet connection is not working?
No, your high-speed Internet connection must be active for you tp make
phone or fax calls.
Can I make calls while I'm browsing the internet?
Yes, you can make phone or fax calls while browsing the Internet.
However, your web browsing may affect the quality of your call, depending on the amount of upstream data traffic.
How do I know If I have a VoIP phone call?
It will ring like any other call on your ATA connected phone.
Can I use my computer while making/receiving calls?
Yes, you can use your computer while making/ receiving calls, for all normal activities including browsing the Internet, if you are using a high speed connection.
Does my monitor need to be turned on?
No, but your Internet connection needs to be working.
Can I have calls to my telephone number transferred to another number?
Yes, the Call transfer feature allows you to do this.
Can I send a fax with VoIP?
Yes, if your provider or equipment configuration allows you to. This is also known as Fax over IP or FOIP.
ATAs usually come with at least 2 connections, one for your phone and the other for your fax. Some problems have been reported with FoIP, mostly to do with the data transfer rate, and in some cases, you may need to configure your fax machine to operate at a slower speed.
Does the SIP protocol support the standard telephone features?
Yes, SIP supports, among others:
- Call Forwarding (unconditional, busy)
- Call Transfer (call control spec)
- Caller ID
- Call Hold
- 3-way Conference and Multiparty Conferencing (call control spec)
- Call Return ("*69")
- Call Park (with Notify)
- Follow-Me
- Find-Me
- Call Waiting
- IVR Systems
- Multiple Line Presence
- Call Waiting
- Camp On
- Call Queuing
- Automatic Call Distribution
- Do Not Disturb
Some services, like repetitive dialing, station speed dialing, last number redial, and distinctive ringing, are implemented purely in the end system and require no support from the signaling protocol.
How does SIP support Caller ID?
Caller-ID is provided by the "From" SIP header containing the caller's name and number. The number would most likely be placed in the "User" field of a SIP URL or appear in a "Tel:" URL.
Since the callee generally does not know or trust the callee's server, only cryptographic signatures can be used to ensure that the information is valid. For example, the outgoing proxy might be operated by an ISP, enterprise or phone company and sign for the identity of the caller, using the "Signed By" parameter, with the identity of the company verified by a public key certificate similar to those used by web sites.
Does SIP carry DTMF?
There are at least two options for carrying DTMF and similar signals in a VoIP network using SIP:
First, DTMF can be transported as an RTP payload (RFC 2833) which has the advantage of providing accurate timing and alignment with the speech RTP packets. Also, media gateways are the most likely to detect and generate tones, so that making it part of the media stream is appropriate. However, under some circumstances, it may be necessary for signaling entities to know about DTMF signals.
Currently, there is no standardized solution within SIP, but it has been proposed to carry DTMF information in SIP "INFO" messages, either encoded as simple text or using the RFC 2833 format. The latter is more complex, but offers duration and timing information. |
