It is sufficient to provide the username or telephone number on a connected institution for calls within your domain (there, where the client is registered). The client automatically fills in it's registered domain and sends it to chosen server. For calls to other domains is needed to provide the whole URI - user@domain. System ENUM can also be used if telephone number in international format is provided (for example 42095001001). The client will execute the command or will fill out the domain and the finding action is let to the server. Plus sign is not required at the start of the telephone number.

Users registered on CESNET SIP server can be dialed by SIP URI with the username or telephone number with a domain. For example, user registration in cesnet.cz domain with user name example and alias 950071001 is available by following SIP URI sip:example@cesnet.cz, sip:950071001@cesnet.cz or sip:420950071001@cesnet.cz and by his telephone number +420950071001 using ENUM or (+420)950071001 using PSTN.

You can test the basic calling functionality on testing telephone number 420950079999 resp. 420950079999@cesnet.cz or 420596991192 resp. 420596991192@cesnet.cz.

SIP server of CESNET2 network is connected with some VoIP operators in Czech republic, which allows a call without a fee in their networks. Numbers of these operators dial in including the prefix 420.

• VoIPex
• 802.vox
• Fayn
• Ha-vel
• LAM - VaseSit
• NETWAY.CZ
• SITKOM

Using SIP URI in format name or telephone_number@domain you can call to some other SIP domains. For example:

• iptel.org - free server of creators of SIP Express Router
• sip.edu - here you can find links about how to call to some European and American universities and Internet2
• SURFNET – Netherland's NREN

Huge enemies of the IP telephony are NAT and firewall because IP addresses and ports where data will be routed are not negotiated until the signalization in a call connection. Most of the modern clients include mechanisms for passage through the NAT. Only some of the types of translations work. CESNET SIP proxy includes another logic which makes this passage easier for clients for the cost of media not going the shortest path but through RTP proxy on SIP server.
Průchod přes firewall je mnohem větší problém. Passage through the firewall is a much huger problem. We recommend using VPN connection for restrictive firewall, although there are situations where even this is not possible.

If you find any problem we will appreciate if you could report it on sipadmin@cesnet.cz address. We need enough information to solve this problem. If the problem is related do telephone calls please state from which URI/number you called and when the call happened. Type of your client would be helpful too. Consider that in some cases are in call involved five and more signalization elements between two clients.

Terms and conditions under which personal data collated in providing the CESNET e-infrastructure services are processed you can find here