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VoIP expert settings

At this point, we provide you with detailed VoIP settings that you need for setting up any SIP-capable devices with an easybell SIP account. However, the default settings of SIP devices are usually sufficient.

We fully support the SIP 2.0 standard according to RFC3261.

Reference settings

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

Registrar: (for the Cloud Telefonanlage

If your PBX does support DNS-SRV you may use the registrar
Please note that DNS-SRV is only provided with this registrar. Both the cloud PBX and encrypted telephony each use different registrars, which is why DNS SRV is not possible in these combinations.

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!


SIP port:

Port for UDP or TCP protocols (default): 5060
Alternate port for UDP and TCP: 5064 (as some routers affect SIP traffic over 5060, which can cause problems)
Port for TLS: 5061

RTP port range in general:  Please generally open ports 10,000 to 50,000 (UDP) for the RTP stream

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

Detailed RTP portranges:

RTP stream for regular VoIP accounts and trunks: 20,000 to 50,000 (UDP)
RTP stream for Cloud Telefonanlage: 10,000 to 20,000 (UDP)
RTP stream for the VoIP to go app: 10,000 to 20,000 (UDP)

RTCP support available: Yes

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

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.

Registrar (IPv4 / IPv6): (For the Cloud Telefonanlage

TLS port: 5061

Port Range: 20000 - 50000 (For Cloud Telefonanlage 10000 - 20000)

SRTP Requirement: SRTP enabled/required

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

Settings for using the 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:

General settings for 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).