Knowledge base A&A SIP / VoIP / mobile technical details
Note: We are deploying a new call server, and so some of the features described are subject to subtle changes.
SIP/VoIP technical details
Our telephone services are designed to be simple to use whether you have a single phone at home, or a multi site office with phones on desks, mobile phones, and hunt groups. The services have a range of features which we are constantly improving. This page details how the features work. Mostly the features are controlled by our control pages and changes take effect immediately.
Numbers
The system is designed around phone numbers just like traditional phones. A login to our phone service is a phone number. Internally we use international format starting + for phone numbers, e.g. +443333400000. However, most parts of the system allow you to use a national format number, e.g. 03333400000.
This means any VoIP equipment you use with our service has to have a real phone number. In simple cases that is just your phone number but in the case of a typical office it means every phone has a direct dial number even if it is also part of a hunt group of some sort. The number does not have to be visible to other people, i.e. calls could appear to come from a main office number for example.
Some features need numbers to work, such as hunt groups, which may not have a phone themselves. We charge for each number you have per month whether you have a phone connected or not. You can however reserve numbers which you can activate when needed, and these are a lower cost while they are reserved. Some really nice numbers cost more.
Equipment
You can connect any suitable equipment to our service. This can be simple SIP handsets, or complete phone systems. Our service makes it simple to run a virtual phone system (centrex) by simply having SIP handsets, allowing internal calls between handsets for free and using simple short number dialling (like extension numbers).
Not all equipment supports all features in the same way. Whilst we will investigate any issues and ensure we meet the standards, we can't guarantee that specific equipment will work. We recommend SNOM handsets as we use them in our own offices and they work well.
Networking, firewalls, and protocols
We support SIP/2.0 protocol using UDP for call control. In most cases we pass SDP/media transparently between customers and carriers. Our carriers and our equipment all support G.711 a-law and you should ensure you configure your equipment to support this. Compression protocols (e.g. GSM) can be negotiated directly between SIP endpoints, but a-law will generally give the best call quality on calls to and from the national phone network. Recorded calls require a-law.
Whilst SIP traffic will always be to/from our call servers the media can be from any source as negotiated in the call set up. The media is normally symmetric ports on UDP, so not an issue with firewalls, but you may wish to allow all UDP to your SIP phones. As ever we recommend not using NAT - we may run a STUN server to assist NAT connected SIP phones.
Our call servers theoretically support IPv6, but we are having problems finding equipment to test against.
- Allow all UDP to/from our call servers (81.187.30.110-119) to your SIP devices for call set up. We will connect from ports 5060-5069 (usually 5060). Your equipment may work on port 5060 or have a narrow range of ports that it uses for call set up which helps limit firewall ranges.
- Allow all UDP to/from your SIP devices for RTP. Most SIP equipment has a range of ports used for RTP and may even be configurable. This allows you to narrow the range of ports used for RTP considerably making more sensible firewall rules. However, RTP can come from anywhere on the internet.
Outgoing features
Outgoing features are those that relate to making calls from a SIP handset or similar equipment.
- SIP registration: You can set up a password to allow SIP handsets to connect to your number and make calls.
- Emergency: Calling 999 or 112 connects to emergency services. This takes priority over centrex or other dialling. These calls are not recorded.
- Local dialling: We can set an area code which is applied if you dial a local number.
- Centrex: We can set a number of digits for short dialling, e.g. 3 digits. If you dial 3 digits then you are dialling the same as your number with the last 3 digits replaced by what you dialled. This is generally used where a company has a block of numbers and other phones in the company are in the same block so dialling a short code gets those phones. This is like dialling an internal extension number on an office phone system. Where the block you have is not the full 100, or 1,000 numbers, then you could dial someone completely different using short codes - there is nothing that restricts the short dialling to just your block.
- Cost limit: We normally restrict calls that could cost more than 20p+VAT for a one minute call. This limit is only 2p+VAT for international calls. Ask sales or support if you need to increase these limits.
- Recording: You can ask for outgoing calls to be recorded and emailed to you. The recording is stereo (one person each side) and can be WAV, MP3 or OGG format. We can even encrypt the recording.
- Presentation number: You can set a presentation number to be used as the calling number when you make calls. This is not used when making centrex calls (short numbers). You can also override this and send your number by prefixing 1470. You have to demonstrate you own the number you wish to use, and there is a set-up fee.
- 2nd Presentation number: We can set a 2nd number which your equipment is allowed to send and we pass on as your number. You have to demonstrate you own the number you wish to use.
- Withhold number: You can prefix 141 to withhold your number when making calls. Bear in mind some people reject such calls automatically.
Incoming features
Incoming features relate to what happens when someone calls your number. You can have numbers that do not have a phone connected, and are used to ring multiple phones.
- ACR: You can set your number to reject calls where the calling number is withheld. The caller gets a suitable message and are not charged for the call.
- Profile: You can set time of day (hour by hour and days of week) where your number is to work or not work. When out of hours you can set an alternative number to ring.
- SIP registration: You can set up a password to allow SIP handsets to connect to your number. When your number is called, registered SIP handsets are called. The system allows multiple simultaneous registered handsets. In general, if you want multiple phones to ring at once we recommend setting up separate numbers for each and set up a group ring.
- Deliver to you: You can set a hostname, username, and password for a SIP system to which we will try and pass the call.
- Group: You can set up a number of additional numbers that are rung at the same time. This is how hunt groups are set up. Each of the numbers can be set to delay ringing so you can make a cascade of phones that ring.
- Divert on fail: A number to ring if your phone (SIP registered) fails for some reason.
- Recording: You can ask for incoming calls to be recorded and emailed to you. The recording is stereo (one person each side) and can be WAV, MP3 or OGG format. We can even encrypt the recording.
- Voicemail: You can set up voicemail, to apply on no answer or busy, and can configure how long to ring before no answer. The recording is emailed to you.
- SIP address: Incoming calls can be made from the internet to sip:number@aasip.co.uk and delivered just like a normal incoming call. The CLI is not trusted for such calls, so are sent to your SIP handset with a ? on the front, and not passed on if the call is diverted.
Note: ACR and Profile apply before considering registered SIP phones and group rings, etc.
Note: An incoming call can be trying to call group numbers, SIP registered phones, and delivery endpoints all at the same time. The first to answer gets the call.
Mobile features
Mobile features relate to using A&A SIM cards.
- SIM/number attachment: You can set a SIM to be associated with a number. This means calls from the mobile are calls from your number. This is very much like a SIP handset registering on the number. All of the normal features of outgoing and incoming calls apply.
- 2nd number: You can associate a phone with a second number as well. This works with a prefix. Calls made from the phone starting 9 are treated as made from this second number. This is very much like most SIP handsets that can register on more that one server at a time. Calls to this second number go to the mobile but with a 9 on the front of the calling number so you can tell it is for the 2nd number. This also helps ensure returning calls dial from the appropriate calling number.
- 3rd number: A 3rd and 4th number can be attached using 8 and 7 prefixes, just like the 2nd number using 9.
- SIP registration: You can set up a password allowing a SIP handset to register as your mobile. This operates with the 2nd number features in the same way as the mobile. This is mainly intended for mobile phones that can also work as a SIP handset, e.g. over wifi, allowing your mobile to work on wifi (and hence cheaper) when at home/office.
- SIP2SIM: Instead of attaching your phone to any numbers you can provide a hostname, username, and password. We will register using those details on a 3rd party SIP server. Calls from the mobile are sent to that server. Calls from that server are sent to the mobile. This allows the mobile to act as an extension off your office phones system, for example.
Diverts and transfers
It is possible to have a SIP handset set to divert or transfer calls. This is treated as a call from your phone.
Diverts and redirects and group ringing work as if dialled from your number, allowing you to use centrex (short number), local, and so on. Note the mobile 9 prefix is a feature related to the SIM and not the number, so cannot be used in diverts and transfers.
Charging
Calls made from your number are charged. Calls are only charged if answered. Calls are charged based on the number called and duration of call (to the second). Different rates apply for peak (9am to 6pm Monday to Friday), off peak, and weekend. We publish the dialling code to rate name, and the cost of each named rate. Not all numbers can be called (e.g. you cannot call premium rate numbers).
Where other numbers are made to ring as a result of a call to your number, such as group ring, out of hours number, failed number ring, divert, or transfer, then they are charged as if called from your number.
Normally one rate applies for the outgoing call based on the number you dialled, but some calls can have more than one charge rate at once.
- If your number is an 0800 (freephone) number, then you have a charge for any incoming calls to that number.
- If an incoming call is delivered to your mobile/SIM then there is a mobile part charge for the incoming call to you. This does not apply if your number starts 07. This only applies if actually routed to the mobile and answered on the mobile, and does not apply if answered via the SIP registration (e.g. wifi).
- If your call originates from your mobile/SIM then there is a mobile part charge in additional to any outgoing call charge made for your call
- Calling an A&A number is normally free of charge. If the A&A number is an 07 number an A&A mobile rate applies even though it is an A&A call as the called party will normally accept the call on a mobile and is not expecting to pay an incoming mobile part on calls to their 07 number.
- The SIP2SIM mobile feature has a charge for the mobile part for all calls to and from the mobile.
- The mobile part charge is different when roaming. Even with an 07 number there may be incoming charges for taking a call on a roaming mobile.
This means you could have two rates applicable at once. E.g. if you had an 0800 number on your mobile you could have two incoming call rates apply (0800 and mobile part). If you had an 0800 number diverted out to somewhere else you could have two charge rates (0800 and the outgoing call rate). Note that if your incoming call to your mobile number is diverted somewhere else then the mobile part does not apply as you do not answer on your mobile.
Your own actions can affect incoming call charges - e.g. if you have a mobile on a normal geographic number, answering on the mobile has incoming call costs (mobile part), but answering on SIP has no incoming call charge.
The action of the person you are calling does not affect the rate(s) applicable to your call. e.g. if they are roaming, or divert their call, or if you call an A&A 07 mobile number it does not matter to your call rate whether they answer on mobile or SIP. Your call charge is based on the number you dialled and not the actions of the recipient in any way (other than the fact they answer the call or not, obviously).
Calls are charged from when they are answered. Recorded messages from the telephone network are not normally charged. Some systems answer then play a message which is a chargeable call. Most phones can indicate if the call has been answered or not ("ringing", or "call" or similar on the display).
Credit control and warnings
We provide real time usage details and call costs via the control pages.
Note: data and roaming may have delayed charges but we aim to update these daily if possible.
Technical notes
- As the system uses phone numbers the domain part is not relevant for registration and incoming SIP calls, but we recommend using the full E.164 number as the username part and sip.aasip.co.uk as the hostname, e.g. sip:+443333400000@sip.aasip.co.uk. We also accept calls to tel: URIs.
- Whilst the system currently uses UDP only, it may be made to use TCP as per the standards if this is required in the future. UDP is preferred.
Call Data Records
Call and text charges are invoiced in arrears. The invoice has a line item for each billable number showing total charges. The PDF attachment includes a printable itemised bill for each billable number. The invoice also has an XML attachment which includes the full CDR for all calls so that you can process these automatically. If you do not have XML billing, ask accounts to enable it for you.
Some calls are just incoming to a phone or outgoing from a phone, but it is possible for a call to come in, and ring multiple other numbers, some of which are A&A numbers that ring multiple numbers, etc. In this case a single call can result in multiple CDRs for separate billable numbers. The CDRs log failed calls if they have got as far as actually trying to call a number (i.e. delayed ring group numbers that have not reached the delay time are not logged).
| Billable number | The CDRs are listed for each billable number. This is either the number of the SIP account or a SIM ICCID. The SIM is used where calls/texts are direct to/from the SIM (SIP2SIM) and no number is assigned. | |
| Call/UID | A unique reference for the CDR entry and a group reference for records that relate to the same call | |
| Call date | This is the date and time the call started. | |
| Call type | Incoming | A call to a number that was not diverted or transferred. |
| Outgoing | A call from a number | |
| Relay | An incoming call that was diverted or transferred to an outgoing number. | |
| Origin | PSTN | A call from the telephone network |
| INTERNET | A call from the internet | |
| SIP | A call from a registered SIP user | |
| SIM | A call from an A&A mobile | |
| SIP2SIM | A call from registered SIP2SIM end-point | |
| AA | A call resulting from an A&A outbound/relayed call | |
| Destination | PSTN | A call to the telephone network |
| INTERNET | A call to a SIP endpoint on the internet | |
| SIP | A call to a registered SIP user | |
| SIM | A call to an A&A mobile | |
| SIP2SIM | A call to registered SIP2SIM end-point | |
| AA | A call to an A&A number | |
| SPECIAL | A call that ended on some special server or announcement | |
| VOICEMAIL | A call that ended on the voicemail system | |
| Status | The final call status as a SIP code. 200 means answered. | |
| Ring time | The total time spent ringing in seconds | |
| Call time | The total chargeable call time in seconds | |
| Calling | The calling number. This is normally the CLI from the first leg in a chain of calls and passed through. This may be withheld or unavailable or untrusted. Untrusted is any CLI from the INTERNET and is prefixed with a ? | |
| Called | The original called number. This is the full phone number from the first incoming leg in the chain of calls. | |
| Dialled | For outgoing and relayed calls this is the number as dialled, which may be a short number or local number, etc. | |
| Target | For outgoing and relayed calls this is the full number that was actually called (based on dialled) | |
| Cost | The price in pence as charged for the call | |
| Calc | The way the cost was calculated | |
| Rate | The name(s) of the rate(s) used to price the call | |
| Clearing | Which end cleared the call | |
| Reason | Where we get it, the reason code on the final BYE (e.g. Q.850 cause code) | |
