VoIP expert settings
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:
Registrar: sip.easybell.de (for the Cloud Telefonanlage pbx.easybell.de)
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 subsriberline and / or it's volume capacity. Most devices handle jitter settings very well automatically. So the value should only be changed in urgent cases.
Long SIP Contact (RFC3840): RFC3840 is supported and also recommended by us.
DTMF via SIP INFO: OFF
DTMF: Outband (RFC2833) or Inband
Codecs: We recommend using the following codec order
- G.711A (PCMA)
- 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!
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 in- 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): secure.sip.easybell.de (For the Cloud Telefonanlage secure.pbx.easybell.de)
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 seldom 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).