Hereafter, we document the features and recommended settings of www.easybell.com trunks in SIP devices.
easybell fully supports the SIP 2.0 standard according to RFC3261.
Reference settings
| Registrar for SIP trunks | sip.www.easybell.com |
| SIP port (default) | 5060 (UDP or TCP) |
| SIP port (alternative) | 5064 (UDP or TCP) |
| RTP port range | 20,000 – 50,000 (UDP) |
| RTCP support | activated |
| RTP keepalive | activated |
| STUN server | deactivated |
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.
| Setting | Recommended |
|---|---|
| Outbound proxy mode | automatic |
| Expired timer | 3600 (min. 600) |
| SIP max forwards | 70 |
| Long SIP contact (RFC3840) | activated |
| DTMF via SIP INFO | deactivated |
| DTMF | outband (RFC2833) or inband |
| Codecs | 1. G.722 2. G.711A (PCMA)* 3. G.711U (PCMU) |
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.
| Setting | Recommended |
|---|---|
| Baud rate | 9600 |
| 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:
| Information | Head 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” |