IP Telephony for your Business

IP Telephony, also known as Voice over Internet Protocol (VoIP), is a telephone connection over the Internet. The network connection beeing used for data transmission is also used for telephony. The transmission of voice packets is prioritized with a mechanism called Quality of Service (QoS) to avoid any disruption of phone calls due to high bandwidth loads. The telephone equipment must be IP Telephony ready. Traditional desk phones (if necessary with adapter) as well as softphones with headsets on PC, Mac and Linux can be used.

SIP is a shortcut for Session Initiation Protocol which is a text-based signalization protocol. It is used for setup and termination of connections between subscribers. The transport protocol RTP (Real-time Transport Protocol) is responsible for the transmission of speech. Together, these protocols are the basis for "Voice over Internet Protocols", ie Voice over IP.

All services and applications based on the Internet protocol (All-IP) are associated to the Next Generation Network (NGN). For all these services and applications a broadband connection serves as transport platform. The most important protocols in NGN are SIP and RTP. Besides VoIP NGN provides other uses, like UMS (Unified Messaging Service), audio and video streaming.

A SIP Trunk is used for connecting Private Branch eXchange systems (PBX) to the NGN. All phone numbers of the PBX are routed via a single SIP account.

• RFC 3261 (SIP) Session Initiation Protocol
• RFC 2327 (SDP) Session Description Protocol
• RFC 1889 (RTP) Real-time Transport Protocol
• RFC 2833 (DTMF) Dual-Tone Multi-Frequency signaling
• RFC 5389 (STUN) Session Traversal Utilities for NAT
• RFC 2543 (HOLD)
• RFC 3515 (REFER)

IP Telephony is possible for any company that has a broadband connection. Each telephone system is suitable for the use of IP Telephony, or can be made ready for use.

In the conversion from ISDN to IP Telephony, there is the possibility of a partial switchover (also called hybrid solution), or a complete switchover by completely replacing the existing telephone line.

During a partial switch incoming calls are routed via the legacy phone line, while outgoing calls are routed via IP Telephony to start saving costs. This is also the preferred solution if the existing telephone contract could not be terminated immediately.

For a complete migration the existing telephone line is completely replaced and inbound and outgoing calls are routed via IP Telephony only. This solution saves even more costs as there are no more high basic fees for ISDN S0 or PMX ports.

blueSIP Premium IP Telephony features:

• SIP Trunk with 10+ voice channels for inbound and outbound phone calls
• free of charge calls between blueSIP customers
• competitive rates for calls to worldwide landline and mobile destinations
• Allocation of phone numbers of all German area codes
• Porting of phone numbers of all German area codes
• Caller ID Presentation or Caller ID Block (CLIP / CLIR)
• Call waiting / Swap / OnHold with Music On Hold
• Fallback routing in the absence of a valid SIP registration
• Callback und Callthrough service
• Fax support with T.38
• Recent call lists (placed and missed calls)
• Voicemailbox with individual greeting message per account
• Up to 100 voicemail messages
• Message waiting signalization to SIP device (MWI)
• 30 days storage time for voicemail messages
• Retrieving voicemail by phone or delivery via email
• Post payment and monthly invoice
• Itemized bill online
• individual charge credit per account
• Fee announcement on/off option (per call or permanent)