FAQ

We use these technologies in many different ways. Sometimes we are not even aware of using it... For instance one may use a calling card number to make a phone call, which routes the call via VoIP gateways. Instant Messaging tools use these technologies. Some people use IP phones (e.g. Vonage) at home. Some use PTT phones which utilize VoIP technology. There are many more areas/ways where we use these technologies. The key areas are illustrated in our eLearning.

Q2:

SIP is a VoIP protocol – so does it carry voice packets?

For the most part SIP does signaling. Voice is carried by other real time protocols such as RTP. There are some implementations however where SIP is used to carry the media itself. For example SIP can carry the media (normally text) of an Instant Message in the payload of a SIP message (request) called MESSAGE

Q3:

Do I always need to use a proxy server?

First of all it is not you, it is rather your SIP phone that may need to use it... Second, the answer is NO. SIP phone needs to use SIP proxy server only when it does not know the IP address (or host name) of the destination or if the policy of the operator (ISP) mandates it. So for example say you like to establish a white board session with a colleague of yours, using SIP based client such as Net Meeting. Now if she provides you the IP address of her machine, you can contact her machine directly.

Q4:

I want to build a SIP phone - is SIP all what it takes?

Well, not quite. If you don’t equip it with RTP stack and couple of vocoders it might be useless. It depends of course on the use you intend for it.

General

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Q5:

What is a SIP phone?

A SIP phone is a SIP user agent that provides the traditional call functions of a telephone, such as dial, answer, reject, hold/unhold, and call transfer. SIP phones may be implemented by dedicated hardware controlled by the phone application directly or through an embedded operating system (hardware SIP phone) or as a softphone, a software application that is installed on a personal computer or a mobile device, e.g., a personal digital assistant (PDA) or cell phone with IP connectivity. Examples include softphones such as X-Lite, and hardware phones from vendors such as Aastra.

Q6:

If I bought a SIP phone, what other hardware or software will I require?

A SIP PBX, Telephony Server or SIP trunk from a SIP provider

Q7:

Will I need external help to configure my SIP phone?

Yes. Although if you buy Aastra Linkpro system along with Aastra SIP phones, the telephones will automatically be configured

Q8:

Can I also call standard telephones and mobiles with my SIP phone?

Yes

Q9:

Does my SIP phone come with a warranty? If yes, what is the warranty period?

Yes, one year

Q10:

If my SIP phone needs repair/replacement or I require support, what should I do?

You may contact support via email or live chat, a support case will be logged.

Q11:

Is there a limit to how many sets I can purchase online?

Q12:

After purchase, in how many days will I receive my ordered set/s?

3-5 working days, subjected to custom clearance.

Q13:

Are SIP phones a dutiable item in my country?

You may need to check this with your country’s current import policies

SIP Functionality

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Q14:

Does SIP 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 conferences and multiparty conferencing (call control spec)
		call return ("*69")
		call park (with NOTIFY)
		follow-me
		find-me
		call waiting
		IVR systems
		multiple line presences
		call waiting
		camp on
		call queueing
		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. The Telecommunications Industry Association (TIA) is working on a recommendation for business PBX-style services and other Internet phone requirements.

Q15:

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 caller 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 signedby parameter, with the identity of the company verified by a public key certificate similar to those used by web sites.

Q16:

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). This has the advantage that it provides 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.

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