SIP trunk phone connections
Up to 200 parallel calls
Phone number without monthly costs
Up to 1 year free test period*
Hosted PBX with numerous features
Easy and intuitive handling
Extensions from 60 cent/month*
Internet & phone for businesses
VDSL connections with up to 100 Mbps
SIP trunk telephone connections
Integrated Cloud PBX on demand
- SIP trunks
- Cloud Telefonanlage (Cloud PBX)
- Business DSL solutions
VoIP Expert Settings
We fully support the SIP 2.0 standard according to RFC3261.
Our standard codec is g.711a, further codecs can only be used if the other side supports this codec.
For an extensive configuration of your SIP system, please use our following reference settings:
Registrar: sip.easybell.de (for use with the Cloud Telefonanlage: pbx.easybell.de)
Protocol UDP or TCP (default): 5060 (alternative port 5064, since some routers affect the SIP traffic over 5060, which can lead to problems)
Protocol TLS (Encrypted Metadata): 5061
Important note: Many terminal devices do not yet support TLS, e.g. FRITZ!boxes. Therefore, you should usually use a TCP port if you do not know whether your device supports TLS.
Outbound Proxy Mode:
Expired Timer: The default value is 3600, the minimum 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.
RTP Portrange: 20.000 to 50.000
RTCP support available: Yes
RTP Keepalive: To ensure that a timeout does not take effect unintentionally: yes
Reference values for:
JB Shrink (Jitter)
Ideally, the min-jitter should be around 20-30, but in principle the jitter should be made more dependent on the line and/or the volume tariff. Most devices handle this very well automatically. The value should therefore only be changed in urgent cases.
Long SIP Contact (RFC3840): RFC3840 is supported and recommended by us.
DTMF via SIP INFO: OFF
DTMF: Outband (RFC2833) or Inband