Connect your Control Desk to the SIP world

Transcription

1 Connect your Control Desk to the SIP world

2 Systems in << << in the VoIP World IoIP over IP Flexible VoIP interface providing SIP/IAX connectivity to integrate standard SIP phones into the system or SIP trunks to 3rd party VoIP servers. Security, high voice quality, additional control and display features, real-time protocol, reliable protocol proven for more than 5 years, wide variety of existing terminals for all kinds of environments, markets and requirements.

3 s integrate SIP SIP VoIP Today s standard for VoIP Internet telephony, based on Internet protocols similar to http web based. Data networks allow transmission of manifold data. VoIP (Voice over IP) is the transmission of speech using Internet Protocol (IP), particularly in telephony. Mobility of phones (soft / hard phones) Easy interfacing Provides multimedia aspects (audio, data, video) in one platform >> >>

4 Simplicity meets Power The Gateway to the World of VoIP Commend s SIP/IAX Interface One of today s mayor topics is the integration of various systems. Just like a decade ago standard telephones were to be integrated in professional solutions, today VoIP telephony systems need a gateway to systems. The idea is to use VoIP telephones or even a VoIP telephone exchange as an extension for systems. Telephones can even be used as sub- or master stations. The key to connect our Servers to a VoIP server is Commend s new SIP/IAX networking card. The use of standard SIP/IAX protocol makes it easy to connect virtually any third party VoIP server and VoIP telephone to the Commend system. Learn more about the IoIP protocol IoIP over IP is a protocol created by Commend especially for professional solutions. Instead of adapting various other protocols and taking into account that some definitions in those other protocols would not be used at all, we created our own protocol. IoIP was designed specificly for applications. It is a real-time protocol including all aspects, like the easy interfacing of third party auxiliary systems. High quality voice communication as well as security and safety aspects were included. On top of being extremely reliable, hazzle resulting from a very common and thus not very secure protocol is excluded. Proven in countless installations all over the world, IoIP has met professional user s expectations. << <<

5 VoIP Technology The technology behind telephone communications has shifted in the past twenty years. Although the touch and feel of telephones still are the same as twenty years ago, the technology behind it all is a new one. Today, most telephone communications take place via data networks speech is converted into data packets which travel through networks. On top of this, plenty of additional services are possible, like transmission of video streams, data and many more. VoIP Voice over IP is a very vast term and includes all technologies to transmit voice as data packets over any type of IP network. Mostly, however, this is reduced to IP-telephony only. Understanding Protocols in general All types of data communication between two devices are described by protocols regardless if data, pictures or voice transmission. In fact, telephones or other devices used for VoIP communications are more or less computers, and before they can talk to each other, they have to negotiate between themselves how the voice communication will take place. There are protocols for signalling which only include the setup of a conversation, which is usually called a session. The most widely used protocol for this is SIP or Session Initiation Protocol. Then there are protocols describing how the data is transmitted during the session. VoIP very often uses RTP (Real Time Protocol). Other standards describe the en- and decoding of voice into data packets, which can be sent over a data network. Only when all parties taking part in a communication agree to use the same set of protocols, they will be able to communicate with each other. It is based on what needs to be transmitted, including multimedia aspects, that the appropriate set of protocols is selected, and manufacturers are free to use existing protocols, to adapt them or to create fully new protocols. >> 5 >>

6 For a global survey Networking of and SIP Servers SIP Trunks SIP telephone exchanges or SIP servers are computers designed to operate as telephone exchange for VoIP telephones using SIP protocol. Any direct connection between two such computers is called a trunk line. Commend s new SIP/IAX interface connects a GE 700 or GE 200 Server to a SIP server via such a trunk line. SIP / IAX Trunk 1 Server n... Channels SIP Terminals Help Points << << VoIP Server Modules SIP IP Phone Soft Client Public Telephone Network IP Phone

7 SIP Phones Also SIP telephones basically are simple computers so called SIP clients. SIP clients (VoIP telephones) are connected with a SIP server (VoIP telephone exchange). Typically more than one SIP telephone need to be integrated into an solution, mostly in a control room situation. For this reason, the easiest connection is to connect the SIP clients via the SIP server on a SIP trunk with the Commend server. IP Phone Help Points Modules Terminals Soft Client Wireless SIP Phone Server >> >>

8 For a great variety of solutions New possibilities for Networking Technical Facts Practical Application Commend s technology is based on a multiprocessor concept. This concept does not have a bottleneck when much activity takes place, nor does it have a single point of failure, where problems with one processor will cause a failure of the whole system. Also separating the connections for IoIP and SIP data streams provides a higher level of operational security. A classical control room is equipped with one or several master stations, which can also be a control desk on which buttons are assigned with signalling or control functions (like showing that a door is open, or activating a door opener, or just programming a direct call on a certain button). Using a graphic user interface (GUI) like COMWIN, the control desk is now on the screen of the computer as well. B IoIP SIP / IAX IP-NETWORK << << G7-DSP-IP G7-DSP-IP G7-DSP-IP SIP / IAX G7-DSP-IP GEP GEN VOIP-SERV C By interfacing with SIP/IAX, it is now possible to use IP-telephones in a remote location (like a call center) as master stations. Here even more a GUI will be very helpful to assist the operators in identifying the location of problems. All stations at various remote locations like parkings can be either master- or substations and are connected with the control room or call center via an IP network.

9 Call Centre Control Room IP Phones Server VoIP Server IoIP SIP SIP Server Server ET 901 PA Help Points Barrier with Module Site 1 Terminals IP Camera Help Points Site 2 Help Points Ticket Machine with Module Site n >> >>

10 Glossary ATA An Analogue Telephone Adapter (ATA) is a device used to connect common analogue telephones to a digital telephone system like VoIP. Asterisk Asterisk is the world s leading open source telephony engine and tool kit. Offering flexibility unheard of in the world of proprietary communications, Asterisk empowers developers and integrators to create advanced communication solutions...for free. Codec A codec is a software part which compresses or decompresses digitalised voice (or video) so it can fit through a limited communication channel such as a dial-up Internet connection. DNS The Domain Name System (DNS) translates between host names (e.g. and their corresponding IP addresses and vice versa. DSP Digital Signal Processor: a micro-processor treating audio data in a digital manner. In intercom systems a DSP can be used for new functionalities like Audio Monitoring or loudspeaker microphone surveillance and it enables OpenDuplex. ENUM ENUM refers to a set of standards and protocols which are used to translate the international dialplan (E.164) between a VoIP network and the -> PSTN network. ENUM heavily relies on -> DNS. Firewall A firewall is a dedicated appliance, or software running on another computer, which inspects network traffic passing through it, and denies or permits passage based on a set of rules. H.323 H.323 is a VoIP protocol used mainly by -> PSTN-providers for audio communication around the globe. IAX The open Inter-Asterisk exchange protocol (IAX) is used primarily by Asterisk -> VoIP Servers to communicate with each other. over IP (IoIP ) over IP is a real-time protocol using IP-networks for networking Servers and connecting Terminals. It features much better speech transmission than VoIP thanks to a higher audio bandwidth of 7 khz. IP Internet Protocol: asynchronous protocol for the packet transmission of information through a common network, e.g. the Internet (www world wide web). ISDN Integrated Services Digital Network: Internationally standardised system for digital telephones which defines both transmission and signalling. LAN Local Area Network: network limited to a small and closed environment, typically within a building, e.g. a company network. Usually a LAN is designed as an IP-network. NAT Network Address Translation Network Address Translation (NAT) is a widely used technique used in routers which translates between an internal network (LAN) and an external network (WAN). A router using NAT is usually open for outgoing traffic but a barrier for incoming traffic. << 10 <<

11 PSTN The Public Switched Telephone Network (PSTN) consists of all of the world s public telephone networks including landline and mobile phone networks. UDP Universal Datagram Protocol: protocol for data transmission without handshake, e.g. used for speech transmission or streaming in networks. QUALITY OF SERVICE (QoS) Procedure in IP networks that prioritises certain connections and / or data packets by granting a defined delay and bandwidth. QoS is also used for the prioritisation of audio data in shared IP networks. Voice over IP (VoIP) VoIP is the overall term for telephony solutions via IP networks. RTP The Real-time Transport Protocol (RTP) is used in -> SIP phones and describes how digitalised and -> encoded audio data is transmitted over the internet. SCCP Skinny Client Control Protocol The Skinny Client Control Protocol (SCCP) is a proprietary protocol used by Cisco Systems for their telephony solutions. SDP Session Description Protocol The Session Description Protocol (SDP) is used in -> SIP phones for the initialization or termination of phone calls. VoIP Server A VoIP Server is a dedicated computer which acts as a private branch exchange providing connections between VoIP and ordinary telephones, the -> PSTN and other VoIP servers, networks and Asterisk. VPN Virtual Private Network A Virtual Private Network (VPN) is a secured, private link though another network. It is commonly used to connect two remote locations over the Internet. WAN Wide Area Network Wide Area Network: network limited to a defined environment, which can include several locations of a company, e.g. several LANs form a WAN. SIP Session Initiation Protocol: network protocol for setup and handle communication session between two or more participants. The protocol is specified in RFC SIP is a frequently used protocol in IP-telephony. Soft Client A Soft Client is a program, running on a personal computer which replaces a common phone by using software and a headset. TCP/IP Transmission Control Protocol: protocol for secure data transmission using IP, e.g. for downloads from the Internet or for s. Handshake and adapts automatically to the available transmission rate. >> 11 >>

12 Technical benefits using Commend vs. SIP telephony solutions and telephony are no competitors. They serve two different purposes, and each one is best suited for its very purpose. While a telephone is used for making direct calls talking into a handset after dialling a certain number, is designed for handsfree conversations, mostly between substations and master stations or control rooms, including many situations where security decisions depend on this conversation. Using the wrong system may have fatal consequences. The benefits of an solution include: Excellent handsfree speech quality with 7kHz audio bandwidth Wide range of stations designed to meet very specific requirements for both installation and design Flexible control of auxiliary systems thanks to a wide range of interfaces Control of Inputs and Outputs for easy control of third party systems Scalable design thanks to a modular system concept DSP Features in real-time PA broadcasts and conferences Integration of analog and digital solutions with IoIP / SIP in one platform Multiple processor design excluding the risk of a single point of failure Proven reliability for security systems in applications all over the world Commend, Inc. 63 Ramapo Valley Road, Suite 201 Mahwah, New Jersey Tel. (201) Fax (201) office@commendusa.com W-SIP-EN SIP Folder Commend v This brochure provides only information of a general nature and cannot form part of a contract. Any information about systems or devices provided in this brochure is subject to technical changes.