SIP Trunk Connection Requirements

To connect to the SkyTel network, your equipment must support standard IP telephony protocols and comply with the following requirements.

Supported Signaling Protocols

The following signaling protocols are supported:

* SIP (RFC 3261)

* SIP-I (SIP with encapsulated ISUP)

Supported transport protocols:

* UDP

* TCP

* TLS

Supported Audio Codecs

The following audio codecs are supported:

1. G.711 A-law (PCMA)

2. G.711 μ-law (PCMU)

3. G.729

G.711 is recommended whenever sufficient network bandwidth is available. The use of other codecs must be agreed upon in advance.

DTMF Transmission

The following DTMF transmission methods are supported:

1. RFC 4733 (RTP Events) — recommended

2. SIP INFO

3. In-band — not recommended, especially when using G.729

Telephone Number Format

Both the Calling Party Number (A-number) and the Called Party Number (B-number) must be sent in E.164 format without the “+” sign and without any technical or routing prefixes.

Examples: 12025550123, 447700900123, 37251234567

For more information, please refer to the E.164 Number Format article.

Caller ID Transmission

SkyTel determines the Calling Line Identification (Caller ID) from the following SIP headers, in order of priority:

1. P-Asserted-Identity (PAI) — recommended

2. Remote-Party-ID (RPID)

3. From

The Caller ID contained in these headers must also be sent in E.164 format without the “+” sign or any technical prefixes.

For more information, please refer to the Caller ID Transmission and Modification article.

CC and CPS Limits

Each SIP trunk has configured capacity limits:

* CC (Concurrent Calls) — the maximum number of simultaneously active calls.

* CPS (Calls Per Second) — the maximum number of new call attempts permitted per second.

Calls exceeding the configured CC or CPS limits may be temporarily rejected. The applicable limits are determined by the service plan or agreed upon individually.

IP Network Requirements

For stable operation and good voice quality, the following network parameters are recommended:

* One-way latency: up to 100 ms

* Maximum acceptable one-way latency: 150 ms

* Jitter: up to 30 ms

* RTP packet loss: less than 1%

Additional Information

Before configuring the SIP trunk, please review the following documentation:

* Technical Conditions

* E.164 Number Format

* Caller ID Transmission and Modification

* CC and CPS Account Limits