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Information about which telephone systems are suitable for your SIP Trunk.

All IP-PBXes which can cope with SIP, the technologies of SIP-DDI and CLIP no screening and Codec g.711 are suitable for easybell Business VOICE.  For HD calling, g.722 is necessary.

There are lots of different IP-PBXes. Please make sure, you take one for your needs. Therefore, you have to think about the needed features. A FRITZ!Box is often fine for SIP trunking, because it is able to deal with up to 20 extensions. To make sure you'll find the best suiting solution, we offer an interactive comparisson of PBXes on our website (in German).

To get the full functionality, we recommend you to use one of our certified IP-PBXes. We made extensive tests in cooperation with the PBXes’ producers to guarantee a full compatibility with our SIP trunks.

You can find different guidances to set up our certified PBXes in our help center.

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The right upload speed is often crucial for trouble-free calls via Voice-over-IP. In this article, you will learn how much bandwidth is required per voice channel, why a security buffer is important and how to calculate the requirements for multiple parallel calls.

For calls to the public telephone network, you should plan for at least 80 kbit/s per voice channel - regardless of whether you are using an IP phone, a local telephone system or a cloud PBX.

The actual bandwidth may vary depending on the codec, encryption and network quality. You should therefore plan for 20-30% additional bandwidth to compensate for fluctuations and parallel data transmissions.

Practical example: For 10 simultaneous VoIP calls, you need at least 1 Mbit/s upload speed.

Our VoIP offer can be used worldwide - from any Internet connection that meets these requirements.

Extra tip: Your broadband connection is not charged for internal calls via a local telephone system.

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You can register five devices at one SIP account. 

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Find out here which audio codecs are supported for internet telephony.

Internet telephony supports all important audio codecs so that the best voice quality can be guaranteed at all times.

  •     G.711 A Law and G.711 U Law - packet sizes of 5, 10, 15, 20, 25, 30, 35 and 40 ms
  •     G.722 (HD) for pure SIP calls
  •     T.38 (Fax)
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The following article shows to what extent you can use SRTP with your easybell connection.

RTP stands for Real-Time Transport Protocol. In VoIP telephony it is used, for transmiting audio signals e.g. the spoken word. Based on this, the Secure Real-Time Transport Protocol (SRTP) additionally encrypts these data streams.

easybell supports SRTP according to RFC 3711.

You can transmit the RTP communication on the route between your terminal device and our servers using SRTP. This means for you that your calls with other easybell customers are encrypted - provided that all participating terminals are correctly configured and support encryption.

If you use our cloud telephone system, you can even have terminals from Snom, Htek and Yealink automatically configured for encrypted telephony via ACS. The necessary settings for manual setup can also be found in our VoIP expert settings.

However, calls that are routed beyond our network boundaries are only accepted unencrypted by the other network operator.

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