Features & reference settings

Hereafter, we document the features and recommended settings of easybell.com trunks in SIP devices.

easybell fully supports the SIP 2.0 standard according to RFC3261.

Reference settings

Registrar for SIP trunkssip.easybell.com
SIP port (default)5060 (UDP or TCP)
SIP port (alternative)5064 (UDP or TCP)
RTP port range20,000 – 50,000 (UDP)
RTCP supportactivated
RTP keepaliveactivated
STUN serverdeactivated

Recommended settings

The following settings have delivered the best results so far for the majority of devices. In specific cases, different settings may lead to better results.

SettingRecommended
Outbound proxy modeautomatic
Expired timer3600 (min. 600)
SIP max forwards70
Long SIP contact (RFC3840)activated
DTMF via SIP INFOdeactivated
DTMFoutband (RFC2833) or inband
Codecs1. G.722
2. G.711A (PCMA)*
3. G.711U (PCMU)  
*G711A provides the highest compatibility and needs to be configured in any case, especially regarding emergency calls!

Jitter

Ideally, the min jitter is around 20-30. Generally, the jitter should be made dependent on the internet connection and the chosen tariff. Most devices manage jitter settings automatically. Therefore, it should only be changed if truly necessary.

FAX devices

The following settings are usually available in all the latest fax machines and have proven themselves in the past in order to achieve the best possible results for FAX over IP.

SettingRecommended
Baud rate9600
ECM (Error Correction Mode)activated
High Speed Fax (Super G3/V.34)deactivated

Signaling incoming calls

For reliable call assignment, you need to know in which format your PBX can route incoming calls. easybell tranfers phone numbers in the format E.164.

In the E.164 format, the international area code is given without leading zeros, e.g. “43” for Austria.

SIP authentication for outgoing calls

To set up a VoIP call, the calling party first sends a SIP invite. In addition to the technical framework, these SIP packages also contain information about the identity of the caller. A separate SIP invite is also generated for call forwarding.

For our infrastructure to process outgoing calls correctly, please use the following scheme:

InformationHead in SIP package
SIP user name in“SIP from address User Part”
CLIP / user-provided Number (UPN) in“From-Display”, “P-Asserted-Identity” (PAI), “P-Preferred-Identity” (PPI) or “Remote Party ID”
Learn more about configuring CLIP no screening in our article Setting up caller identification (CLIP).
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