10   Voice Services in CESNET2 Network

10.1   Introduction

The project started in the second half of 1999. Its goal was testing usability of technologies for the voice and data network convergence. Since the beginning, the project directed its attention at the area of Voice over IP (VoIP). In retrospective, this direction has proved correct: alternative ways such as Voice over ATM and Voice over Frame Relay were less successful. After first two years, the value of this project increased significantly. Results of the experiments were successfully tested in operation and the project was moved among the strategic projects.

In 2001, the VoIP network was interconnected with the public switched telephone network (PSTN) which caused further expansion of the IP telephony network: by the end of 2003, most CESNET members participated in the project. Interconnection with the PSTN network allows significant price reduction of telephone charges thanks to low interconnection fees at a quality comparable with classic calls using the PSTN.

This project is building an advanced experimental platform usable also by other research projects; it also allows the CESNET2 network to support this progressive voice communication method. A general project goal is providing the Association members with support for connecting and using the CESNET IP telephony infrastructure. The research goal consists of IP telephony component and application certification and development.

Project outputs are tested in operations, regularly published and presented at conferences. In the area of international cooperation, VoIP interconnection with foreign institutions is being tested; we also play an active role in a TERENA project - the IP Telephony Cookbook publication.

10.2   Changes in the VoIP network infrastructure

Other Association members joined this project during 2003. New Voice Gateways (VoGW) were installed in the following network locations:

The following connectivity changes took place in 2003:

In 2003, PBXs of all institutions cooperating in this project use only the ISDN technology. In the early stages of this project, other interconnection modules (E&M, FXS, FXO) were also used. The Silesian University in Opava and the Physical Institute in Prague had taken an advantage of an alternative connection. In 2001 we set up requirements for connecting PBXs into the infrastructure whereby only the digital connection using ISDN was preferred. This allows sending the caller identification which is important for charging calls, collecting statistics and tracing malicious calls. Currently, by the end of 2003, all PBXs of Association members satisfy the rules given in the project cooperation contract.

A significant change achieved this year is an upgrade of the PSTN conectivity through the GTS Czech operator. This connection uses 2×ISDN/PRI connected through a new CISCO 3745 voice gateway. Calls can be directed to PSTN via the Aliatel or GTS operators. This allows better flexibility and increases the service accessibility. The interconnection with Aliatel via NIX.CZ uses the IP protocol.

10.3   Current state of the IP telephony network

Over ten thousand participants from those Association members whose PBXs are connected via voice gateways (VoGW) use the IP telephony service in 2003. The VoGWs and PBXs are interconnected using ISDN; this allows keeping the call detailed records (CDR) in an SQL database at a RADIUS server.

Since its establishment, the CESNET IP telephony network has been oriented towards using the H.323 protocol (minimum version 2); internal network components (Gateways and Gatekeepers) are based on the Cisco platform. Its advantage has been proved during network upgrades and management. Connectivity of the CESNET Association non-members is provided by an external Gatekeeper gk-ext.cesnet.cz located in Prague and running on a Linux platform. From the hierarchy point of view, this Gatekeeper is linked above the internal Gatekeepers in Ostrava and Prague.

The internal Gatekeepers (GK) back up each other; all CESNET network VoGWs can connect to both GKs using different priorities dependent on their geographical positions. Ordinarily, the higher priority connection is active. Requests to connect outside the CESNET2 VoIP network are directed to the operator providing public telephone services; these calls are charged according to the price list which is a part of the contract between the CESNET Association and the connected organization. The network logical diagram is shown on Figure.

[Figure]

Figure 10.1: Logical diagram of the IP telephony network

Calls are charged using 1-second increments; no minimum call duration is charged. Calls within the CESNET2 network between the VoIP project members are free of charge. The call fees are calculated using the TAS-IP application; the IPTA program developed by CESNET employees is used experimentally. Call details are recorded in databases of both applications.

Institutions listed in table are registered on internal GKs in Prague and Ostrava. They can access the PSTN through the Aliatel GK remote zone or through the GTS VoGW.

InstitutionPrefix Phone number
Czech Technical University,
Prague Institute of Chemical Technology,
CESNET9422435xxxx
Technical University Ostrava#059699xxxx, 59732xxxx
Masaryk University Brno754949xxxx
University of South Bohemia Č. Budějovice85838777xxxx, 38903xxxx
Charles University Prague, Rectorate98224491xxx
Charles University Prague, Faculty of Paedagogy94221900xxx
University Pardubice22466036xxx, 466037xxx, 466038xxx
University Pardubice in Česká Třebová83465533006, 465534008
Technical University Liberec40, 4748535xxxx
Faculty of Pharmacy Hradec Králové55495067xxx
University Hradec Králové56495061xxx
Institute of Economy Prague#0224095xxx, 224094xxx
Institute of Economy in Jindřichův Hradec#0384417xxx
Silesian University Opava*0553684xxx
Silesian Univ. Karviná, School of Business Admin.9596398xxx
Academy of Sciences Prague0**26605xxxx
Technical University Brno0*854114xxxx
Palacky University Olomouc0*858563xxxx, 58732xxxx, 58744xxxx
Purkyně University Ústí nad Labem-47528xxxx
University of West Bohemia Pilsen#037763xxxx
University of Ostrava79597460xxx, 596160xxx

Table 10.1: Member institutions in the IP telephony project

Institutions listed in the following Table are registered on the CESNET external GK; they cannot access the PSTN. However, communication between institutions registered on the internal GKs and the external GK is not limited. During 2003, SANET institutions were included in the CESNET H.323 external peering. As a result, one can call, e.g., the Slovak Technical University and Žilina University free of charge:

InstitutionPhone number
CERN (www.cern.ch)00412276xxxxx
FERMILAB (www.fnal.gov)001630840xxxx
SLAC (www.slac.stanford.edu)001650926xxxx
STU Bratislava, Rectorate00421257294xxx
STU Bratislava, Fac. of Mechanical Engineering00421257296xxx
STU Bratislava, Fac. of Civil Engineering00421259274xxx
STU Bratislava, Fac. of Material Science and Technology00421335511xxx
Žilina University0042141513xxxx

Table 10.2: Foreign institutions registered on gk-ext.cesnet.cz

Figure shows the IP telephony network by the end of 2003

[Figure]

Figure 10.2: Internal diagram of the CESNET IP telephony network (large image)

Fees for calling the PSTN are advantageous because all external voice traffic is terminated in a single point in Prague and the telecom operator's interconnection costs are low. Some example costs in 2003: Prague 0.82 CZK/min in peak hours, 0.51 CZK/min off peak, Germany 1.97 CZK/min, USA 2.01 CZK/min. Instructions on IP telephony use are available at http://www.cesnet.cz/iptelefonie/voip-cesnet.html

10.4   Rules for cooperation in the IP telephony project

Organisations connected to the CESNET NREN and conforming to the technical requirements may cooperate in the IP Telephony project.

10.4.1   Technical requirements

10.5   Applications for charging calls

Call records are sent to RADIUS server to be used by the IPTA and TAS-IP applications. The first planned modification undertaken concerns the IPTA application: support for the SIP protocol calls (which requires processing postfix identificators using e-mail format, in addition to prefix identificators formatted as telephone numbers). Some 50 % of this task have been completed.

The second accounting application TAS-IP which was bought by the end of 2003 is also running routinely.

[Figure]

Figure 10.3: Selection mask of institution and call type in the TAS-IP application

The Figure shows a mask used for selecting an institution and an appropriate type of call. The application allows generating standard summaries as well as detailed printouts for selected time periods. Figure shows monthly summary printouts of several selected institutions. The topmost part includes a monthly summary of calls within the CESNET2 network (these calls are free of charge). The bottom part shows a detailed printout of all calls from voice gateways including the total connection time.

[Figure]

Figure 10.4: Example of monthly summaries for selected institutions

Volumes of VoIP traffic within the CESNET network are depicted on the Figure which shows the development during the last two years. New university PBXs are being connected and traffic will continue to grow. The figure shows a substantial growth during 2003. Comparison of latest data, November 2003, with those dated November 2002, shows a sixfold traffic growth.

[Figure]

Figure 10.5: Mothly traffic volumes since January 2002 (large image)

During our project development, two protocol directions arose which are being solved in the project research section. Specifically, these are the H.323 protocol (based on the ITU-T standard for multimedia communication using data packet networks), and the IETF SIP/SDP protocol. Both standards are important in the VoIP field and the project team must pay appropriate attention to both of them.

10.6   The H.323 area tasks

The H.323 signalisation network has been deployed successfully and it operates routinely now. Connecting the voice gateways is being offered to the CESNET members as a part of project cooperation. A part of the project included further tests of IP phones; this is still an experimental field.

10.6.1   Testing the H.450 services

In the area of H.323, tests of H.450 services have been completed: we attempted to test as many H.323 services as possible from all twelve defined in the current ITU-T H.450 recommendation. Tests were conducted using the Siemens OptiPoint400 Standard and Welltech LanPhone 101 phones. We found out that the Lan Phone 101 supports services H.450.1 to H.450.4 while the OptiPoint 400 Standard supports services from H.450.1 to H.450.4 plus H.450.7. A list of tested H.450 services follows:

[Figure]

Figure 10.6: The OptiPoint400 Standard IP phone

10.6.2   H.323 IP Phone and NAT

Another task included testing H.323 telephones on private addresses behind a Network Address Translator. The OptiPoint400 Standard and Welltech LanPhone 101 IP phones were used again.

[Figure]

Figure 10.7: The LAN Phone101 IP phone

The Welltech LanPhone 101 gave better results: it allows configuring a shared public IP address in the "IP sharing" item. The IP telephone has a private address from the range of the network behind the NAT, of course, but after the shared public address is configured, the phone inserts this public address into the signalling messages. As a result, no H.323 support is required from the NAT.

Configuration command

ifaddr -ipsharing 1 195.113.113.151

sets the shared address which the phone should use. On the other hand, the OptiPoint400 Standard requires the H.323 NAT support. For this purpose, routing the TCP ports 1710-1720 and UDP ports 5010-5017 to internal addresses was sufficient.

10.6.3   The Kerio Technologies Products

This Project team has been involved in long-term cooperation with the Kerio Technologies company, a developer of VoIP program applications. Project members test the Kerio products and in return, they are allowed to use these application within the CESNET2 network.

In the first half of 2003 we have tested the Interactive Voice Response (IVR) system which could be used, e.g., as a base for information call centre or as an information system. We created necessary scripts using the VoiceXML language which reproduce a selected text to users using voice synthesis. Callers can use the DTMF tone dialing to traverse the information tree; automatic switching to a phone number included in the script has been tested successfully, too.

Kerio Technologies suspended development and sale of its VoIP products in September 2003. Our external H.323 peering is based on a Kerio Gatekeeper (a Linux application); therefore, another suitable platform should be found to replace it by the end of 2004.

10.6.4   New version of H.323

In July 2003, the ITU-T H.323 version 5 recommendation was released officially. Considering the number of new features, we are glad that the H.323 standardising process is still active. The fifth version brings new management functions and new services; in adition, it is backwards compatible with previous versions.

10.6.5   H.323 interconnection with the CESNET2 network

Two possible interconnection modes exist: direct routing to VoGW, or through the central Gatekeeper (GK). Fundamental difference lies in the H.225.0 signalling mode which has two parts - RAS signalling and Q.931 (Call Signalling). If direct routing to VoGW is used, RAS signalling is not used - see the Figure.

[Figure]

Figure 10.8: H.225.0 signalling routes

Our preferred solution uses the CESNET GK gk-ext.cesnet.cz. The connection methods and actual configuration files can be found in the CESNET Technical report 28/2003.

10.7   Tasks solved in the SIP area

In 2003 we continued testing the infrastructure components which use the SIP signalling protocol.

10.7.1   SIP Server and gateways

Core component of the SIP VoIP network is a proxy and registrar server SIP Express Router (SER) installed on a PC platform under a Linux RedHat operating system. At present, this server can process requests of directly connected test SIP clients and redirect calls to all VoGW of connected institutions. These VoGWs, used in the H.323 network, can process concurrently both the SIP and H.323 signalling protocols.

Rules for call forwarding to voice gateways according to telephone number prefixes are stored directly in the configuration file. Therefore, changing these rules without restarting the server is impossible. We are considering building an external routing data storage but this solution would require writing a new server module as well. This function is not necessary at present and so its implementation has been postponed.

Rather a more crucial problem is the inability to establish a SIP connection from PBXs of affiliated institutions. Users dial a special prefix, e.g., 94, to connect using the VoIP network. As a result, the call is connected using the H.323 signalling protocol. Creating another prefix for entering the SIP network would be absolutely inconvenient: it would only confuse the users and it would not improve the functioning and comfort of the VoIP service.

One possible solution would be using the ENUM services - translating phone numbers to URI using domain name server records: after receiving appropriate ENUM records from the name server, the gateway should be able to decide which signalling protocol is to be used for the connection. Unfortunately, the ENUM services do not yet operate perfectly on the gateways; we endeavour to solve this issue by cooperating closely with the gateway manufacturer.

To demonstrate the SIP routing configuration, a part of the ser.cfg is presented. This determines the routing functions of the SIP Express Router (SER) application working as a proxy and registrar server. Called numbers can be routed using an international format starting with 420 or using the nine-digit national format.

# main routing logic 
route { 
    if (uri==myself || uri= "[@:]cesnet\.cz([;:].*)*") { 
        # Save location contact 
        if (method=="REGISTER") { 
            if (!www_authorize("cesnet.cz", "subscriber")) { 
                www_challenge("cesnet.cz", "0");
                break; 
            }; 
            save("location"); 
            break; 
        }; 
        # Gateway forwarding section start 
        if (uri= "^sip:(420)?22435[0-9]{4}@") {
            rewritehostport("GWIPaddress:GWport"); # CVUT Praha 
            route(1); 
            break; 
        }; 
        if (uri= "^sip:(420)?59(699|732)[0-9]{4}@") {
            rewritehostport("GWIPaddress:GWport"); # VSB Ostrava 
            route(1); 
            break; 
        }; 
        if (uri= "^sip:(420)?54949[0-9]{4}@") {
            rewritehostport("GWIPaddress:GWport"); # MUNI Brno 
            route(1); 
            break; 
        }; 
        #. 
        #. 
        #. 
        # Other gateways 
        # Gateway forwarding section end 

        #  native SIP destinations are handled using our USRLOC DB 
        if (!lookup("location")) { 
            sl_send_reply("404", "Not Found");
            break; 
        }; 
    }; 
    # forward to current uri now 
    if (!t_relay()) { 
        sl_reply_error(); 
    }; 
}

Further important function of the core SIP server is the authentication and the authorisation of IP phones and software clients. The server is equipped with modules which provide these functions by querying a MySQL database or a RADIUS server. However, most authentication and authorization functions within CESNET are resolved using the LDAP directory services. We managed to acquire the server LDAP modules from the development team and we are testing them. Unlike the core of the server, these modules are not available under the GPL license. As soon as the tests of AAA mechanisms are finished, accessing the PSTN via SIP clients will be possible.

Full integration of the SIP and H.323 networks requires an operating translation gateway of signalling protocols. The products we started testing by the end of last year proved unsuitable; therefore, tests of an Asterisk software PBX are being prepared.

10.7.2   SIP IP telephony clients

Here we concentrate particularly on testing the IP phone clients. In the field of hardware IP phones, we are concerned especially with the Cisco and Siemens products. An advantage of Cisco IP phones is the fact that an easy change of firmware allows connecting them to the Cisco CallManager. For the Siemens IP phones, three firmware versions are available: SIP, H.323, and HFA.

In addition to tests of hardware phones, we attempt to find a suitable software client for two favourite platforms - Linux and MS Windows. One of the agents for the Windows platform is the Windows Messenger (not to be confused with the MSN Messenger), currently in version 5. This version corrects major problems in signalling protocol implementation. This client allows voice calls, videoconferencing sessions as well as sending short text messages. A Linux client with similar features is the Wirlab Kphone, currently in version 3.14 - see Figure.

[Figure]

Figure 10.9: The KPhone SIP client

We also test clients made by SJLabs, Xten and others. Our main requirements are stability, quality of user interface, sufficient choices and quality of codecs and a sufficient number of functions provided.

10.7.3   Interconnecting the SIP IP telephony networks

One of the criteria of a telephony network quality is the number of reachable workplaces. To localise the called participant, SIP uses the DNS service. Calls into and from outside the CESNET Association networks can be realised without additional settings or difficult management of special connecting rules. Using the SIP protocol, one can call the iptel.org, SANET, NASK, MIT and many other networks. The experimental service ENUM (RFC 2916) provides translation between the E.164 numbers and URIs. As a result, an E.164 number acts as a unified identifier for accessing several different services (H.323, SIP, e-mail, Web, etc.).

10.8   Further project goals

The team members prepare a set of rules of IP phone operation for members of the CESNET Association. These are intended especially for remote workplaces with an IP connectivity where few phone connections exist and where IP phones could replace the fixed phone lines. In present, IP phones are used mainly by the CESNET staff. A mass deployment of IP phones is expected in the second half of 2004.

To make the VoIP service extensible and manageable, the project team also prepares rules for using the IP phones to which applicants for an IP phone service with a public phone number must conform. The public phone numbers for IP phones have been allocated to CESNET by the GTS Czech operator. We are also negotiating with the Czech Telecommunications Office to be allocated an access prefix for the CESNET2 network.

The project team has the following priorities:

10.9   Project publishing activities

We participate in the TERENA international project IP Telephony Cookbook; our contributions become the Cookbook subchapters. Mr. Sven Ubik is the CESNET contributor. The project results are published regularly and presented at conferences.

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