FAQ Technical

What do I need to use the Voxbone service ?

=> You need a SIP compatible equipment with a public IP address and you need to be capable of receiving calls without being registered. Voxbone does not provide SIP registration nor NAT traversal service.

I do not have a fixed public IP address and I only have a SIP equipment, can I still use the Voxbone service ?

=> It might work, however Voxbone will not support users behind NAT and with dynamic IP addresses.
If this is your situation, let us know and we will refer you to one of our resellers near you.

How do we need to configure our equipment to interoperate with Voxbone ?

=> Sample configurations for all compatible and tested equipments can be found under the Knowledge Base on www.voxbone.com.

What is SIP ?

=> SIP stands for Session Initiation Protocol, defined in RFC 3261. It is the most widely used standard for VoIP signalling. It was designed to be a general-purpose way to set up real-time multimedia sessions between groups of participants. SIP is used during and after the call set-up.

What is a SIP URI ?

=> URI Means "Uniform Resource Identifier". A SIP URI is used to identify an entity to be addressed via SIP, just like an e-mail address identifies the e-mail recipient. The general format of a SIP URI is user@domain:port
"user" identifies the particular user to address. The user can be a phone number, if this is supported by the SIP server.
"domain" can be a fully-qualified domain name or a numeric IP address.
"port" is optional, and if not specified then the standard value of 5060 will be used.
Examples of valid SIP URIs: 11223344@sip.voxbone.com or user1@10.10.10.10:5090

When I map a number to a URI, is the mapping done in real time ?

=> Yes, all configuration changes are done in real-time.

What is a VoIP codec ?

=> The word "codec" comes from COder/DECoder. A codec defines the algorithm used to convert analog signal into digital signal and vice versa. Each algorithm consumes different amount of bandwidth and provides different voice quality level. In the VoIP world the codecs are used to encode the voice for transmission across an IP backbone. Another name for voip codecs is "vocoders".

Which codecs do you support ? (to be updated)

=> We support the standard codecs G.711a, G.711u and G.729, and offer them in that order.

How do you carry the DTMF tones ? (to be updated)

=> We carry DTMF as RTP payload (see RFC 2833). This is the standard way to carry out-of-band DTMF via SIP. It has the advantage that it provides accurate timing and alignment with the speech RTP packets.

Do you provide the CLI in the calls ?

=> Yes, if the CLI is provided to us from the PSTN, we forward the it unchanged. We also clean local CLIs into E.164. This option can be configured online. CLI is generally guaranteed for national calls.

Why do we need to authorize several IP addresses ?

=> This is for redundancy and failover reasons. As we have a completely redundant infrastructure the calls might come from any of these IPs.

What is E.164 ?

=> E.164 is an international numbering plan for public phone systems in which each assigned number contains a country code, an area code, and a subscriber number.
There can be up to 15 digits in an E.164 number.
The E.164 plan was originally developed by the International Telecommunication Union (ITU).

Example of phone numbers written in E.164 format:
12121234567 (country: USA - 1, area: New York - 212, local number: 1234567)
442071234567 (country: UK - 44, area: London - 207, local number: 1234567)
81312345678 (country: Japan - 81, area: Tokyo - 3, local number: 1234567)

I just want to send the calls to our softswitch IP address. What do I need to do ?

=> You need to add your softswitch IP address in the "Endpoint URIs" menu and then map your numbers to your URI.
For example if you just want to receive all the calls in E.164 format what you can do is the following:
STEP 1. In the "Endpoint URIs" menu create the following SIP URI: {E164}@X.X.X.X (where you replace X.X.X.X by your softswitch IP address).
STEP 2. In the "Forward number" menu choose the SIP URI defined in STEP 1.

From which IP address will the calls originate ?