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General information

The Hungarian Research and Educational Network operated and developed by the NIIF Institute has evolved to be able to deliver the most advanced  communication services. The VoIP (Voice-over-IP) service, which provides telephony over IP networks, has been launched in 2003. VoIP is a flexible and cost-effective alternative for our member institutes to solve their communication problems. The call between two member institutes is routed on the Hungarian Research and Educational Network, the HBONE. Call towards public networks terminated by a hungarian telecommunications operator,  selected yearly via a competitive process.

Structure of the VoIP network

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Figure 1: The VoIP network of the NIIF

The sematic structure of the NIIF VoIP network is shown on the figure following.

Institutions with high speed network connection (at least 34 Mbps) to HBONE can join the VoIP network of the NIIF. The calls coming from these institutes first arrive in the central call router. From here, the call can be diverted to:

  • Another institution via HBONE,
  • Another institution in a foreign country via GÉANT research network,
  • PSTN operators via VOIP peering

Connection of an institution

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Figure 2: Connecting institutions

The physical connection of an institution is shown on the following figure.

If there is a traditional private branch exchange (PBX) operated at the institution, the calls are transferred from the PBX to a VoIP gateway using ISDN interfaces (mainly ISDN PRI). The gateway digitizes then packs the voice into IP packets, then sends them to the central call director through the HBONE, using SIP signaling protocol. Cisco 2651, AS5350, Cisco 2811 or 7200 routers are used for ISDN PRI, Cisco 1751/1760 routers are used for ISDN BRI.
Institutions having their own IP PBX equipment (Asterisk, Cisco Call Manager) can join directly the NIIF VoIP network. In this case there is no need for an extra VoIP gateway, all of the VoIP-related tasks (encoding voice traffic into IP packets, transferring) can be done by the IP PBX equipment of the the institution. The communication between the central call director and the institution's IP PBX equipment are managed using SIP or H.323 signaling protocol.

PSTN/PLMN connections

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Figure 3: PSTN connections

The NIIF VoIP network connects to the contracted PSTN service provider at one physical point redundantly, in the headquarters of the NIIF. The aggregated calls routed to the public network are handed over using redundant ISDN PRI connections to the service provider. Calling from a PSTN to the NIIF VoIP network is currently unavailable, due to administrative restriction of the Hungarian law. The service provider's ISDN connections are ended in two Cisco 7200 routers. The central call director uses SIP to communicate with gateways managing service provider's outputs.

If there is a traditional private branch exchange (PBX) operated at the institution, the calls are transferred from the PBX to a VoIP gateway using ISDN interfaces (mainly ISDN PRI). The gateway digitizes then packs the voice into IP packets, then sends them to the central call director through the HBONE, using SIP signaling protocol. Cisco 2651, AS5350,, Cisco 2811 or 7200 routers are used for ISDN PRI, Cisco 1751/1760 routers are used for ISDN BRI.
Institutions having their own IP PBX equipment (Asterisk, Cisco Call Manager) can join the NIIF VoIP network. In this case there is no need for an extra VoIP gateway, all of the VoIP-related tasks (encoding voice traffic into IP packets, transferring) can be done by the IP PBX equipment. The communication between the central call director and the institution's IP PBX equipment are managed using SIP or H.323 signaling protocol.

Peering connections

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Figure 4: Peering connections

The central call director used in NIIF VoIP network can accept peering connections. H.323 and SIP signaling protocols are both used for communication between networks. In case of H.323 calls, the gatekeeper function is provided by central call director.

Central call director

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Figure 5: Routing calls in the central call director

Routing calls in the VoIP network of HBONE is managed by the Deverto Tequet softswitch which is located in the NIIF Headquarters. What does this softswitch do? It accepts calls from the institutions, checks their permissions and forwards calls to their desired destination. Controlling and accepting calls are always managed centrally; every single control message processed by the softswitch.

The network is based on Session Initiation Protocol (SIP). The service provider's and the institutions' line terminator ISDN-SIP gateways are connected to the soft switch using SIP signaling protocol. For institutions who have VoIP PBX, have the possibility to connect to the HBONE VoIP network without gateways. In such cases, number validation and transformation tasks are added to the function list of the soft switch. In every other cases, these tasks are performed by the gateway. Using Multi Protocol Layer Switching Virtual Private Networks (MPLS VPN) is a known method to protect an institution's VoIP PBX. The soft switch installed in the core of the HBONE+ network can provide routing amongst MPLS VPNs. Routing calls between institutions is ruled by ENUM DNS entries: the phone numbers, phone number intervals are represented by DNS entries. The soft switch sends queries to the DNS servers to determine the routes to the desired phone numbers. In case there is no DNS entry for the dialed phone number, it is not possible to reach that number in the HBONE VoIP network. So the call should be diverted to one of the peering partners. In this case the soft switch examines if the dialed number is available for free (via peering) and tries to set up the call. Lastly, if there is no free way to call the number, and the caller has the proper permission for it, the soft switch passes the call to the PSTN/PLMN service provider. Currently, two peering partners are connected to reach each other's phone endpoints for free: the researchers of CESNET (Czech Republic) and GRNET (Greece). So, calls not only arriving, but coming from these networks to their destinations (institutions) are routed by the softswitch.

In the NIIF VoIP network, the central call director generates the billing information and transfers it to a redundant accounting database. Additionally, all of the gateways in the VoIP network send Radius accounting information to the Radius server (Cisco AR). The Radius messages don't count in call data gathering and generating accounting data, they are only for maintenance needs.

Current state of the network

In the VoIP network of the NIIF has 72 member institutions (10 of these have IP PBX : 3 Cisco Call manager and 7 Asterisk). Invitel, a Hungarian telecommunication service provider, terminates calls towards public networks. Outside of Hungary, we have connections to the VoIP networks of Croatia, Czech Republic, Greece and Portugal. Making more peering connections is in progress.
Institutions, which are connected to the NIIF VoIP network, are shown on the map of Hungary.