We fully support the SIP 2.0 standard according to RFC3261.

For a high-level configuration of your SIP setup, please use our reference settings below:

Reference settings


The registrar depends on which product or which additional function you want to use. The default registrar for VoIP accounts is sip.easybell.de.

You can find all other easybell registrars in the following list.

Registrar Product DNS-SRV TLS / SRTP
sip.easybell.de SIP Trunk / VoIP no no
secure.sip.easybell.de SIP Trunk / VoIP no yes
voip.easybell.de SIP Trunk / VoIP yes no
pbx.easybell.de Cloud Telefonanlage yes yes

Please note that for the functions DNS-SRV and encrypted telephony, different registrars must be used in each case. A combination of both functions is therefore currently only possible with the cloud telephone system.

Important note for redundant internet connections:

If you use more than one internet connection at the same time, a registrar with DNS-SRV functionality has to be used. Otherwise, there may be problems with the registration or audio problems.


 UDP / TCPUDP / TCP (alternativ)TLS

SIP trunks / VoIP

Cloud Telefonanlage50605061


Some routers can cause problems by affecting SIP traffic over 5060. For phone numbers and blocks (not the Cloud Telefonanlage), you may use the alternate port for UDP and TCP 5064

RTP Port ranges

If you want to ensure proper transmission of the RTP voice stream for all our voice services, you can enable ports 10,000 to 50,000 (UDP), 

If you want to enable a narrower range, please note the following:

More detailed breakdown of the RTP port range

Type of VoIP accountportrange

SIP-Accounts and SIP-Trunks

(single numbers and blocks of numbers)

20.000 - 50.000 (UDP)
Devices of Cloud Telefonanlage (Hosted PBX)10.000 - 20.000 (UDP)

VoIP to go-App

(For both SIP accounts and cloud phone system)

10.000 -20.000 (UDP)

RTCP support available: Yes

RTP keepalive: Yes (To make sure that a timeout does not interfere unintentionally)

Additional ports for the VoIP-to-go app: 4998 (UDP) & 4998, 5000, 4210, 4280 (TCP)

Additional ports for Cloud Telefonanlage: 4998 (UDP) & 4998, 5000, 4210, 4280 (TCP)

Encryption of meta and media data

Encryption of metadata (SIP), as well as media data (RTP) for inbound and outbound telephony is only usable if your device supports TLS and sRTP (both are mandatory). See our help section for more detailed instructions on setting up encrypted telephony on your VoIP device.

Please note that connections with the VoIP to go app cannot be encrypted at this time

  voip accounts Cloud Telefonanlage
Registrar (IPv4 / IPv6) secure.sip.easybell.de pbx.easybell.de
TLS port 5061  
Port Range 20000 - 50000 10000 - 20000
SRTP Requirement SRTP enabled/required SRTP enabled/required

Advanced settings

STUN server: Please do not use a STUN server!

Outbound Proxy Mode: Automatic

Expired Timer:The default value is 3600, the minimum is 600.

SIP Max Forwards: For fail-safe telephony, easybell has increased the SIP Max Forwards compared to other providers. The default value here is 70.

Reference values for JB Min, JB Max & JB Shrink (Jitter): In the ideal case, the min jitter is around 20-30. In general, the jitter should be made dependent on the internet subscriberline and / or it's volume capacity. Most devices handle jitter settings very well automatically. So the value should only be changed in rare cases.

Long SIP Contact (RFC3840): RFC3840 is supported and also recommended by us.


DTMF: Outband (RFC2833) or Inband


Codecs: We recommend using the following codec order

  1. G.722
  2. G.711A (PCMA)
  3. G.711U (PCMU)

But in any case, the codec G.711A must be configured. Since this codec is the minimum consensus on which was agreed in Europe. By using G.711A, any participant of a call should be able to handle this codec. This is an essential setting in the context of emergency calls!

easybell VoIP to go App

In rare cases it may be necessary to consider the following points of your firewall settings or router configuration, if you encounter problems using the easybell VoIP to go App.

The following ports are responsible for SIP traffic and must be open:

TCP: 4998, 5000, 4210, 4280

Please make sure the following port range is open for the application's RTP packets:

UDP: 4998, 10000-20000

The host to which the application logs on is:


FAX devices

The following settings can usually be found in any, modern fax machine. The settings shown here have been used in the past to achieve the best results when it comes to fax over IP.

Baud rate: 9600

ECM (Error Correction Mode): enabled

High Speed Fax (Super G3/V.34): disabled

For general information on fax transmission via VoIP, see our detailed article on fax (in german language).