Technical conditions of client Internet access for connection to SkyTel network via SIP

Connecting to the SkyTel network via SIP (Session Initiation Protocol) requires meeting certain technical conditions to ensure stable and high-quality voice and data transmission. Below are the main requirements for the client's internet access technical specifications when connecting to the SkyTel network via SIP.

1. Stable Internet Connection

To ensure proper SIP operation, a stable internet connection is required. The connection requirements include:

• Bandwidth: The minimum internet speed for voice calls is 128 kbps per voice channel (both directions). For video calls, the requirements may be higher, depending on the resolution and codecs used.
• Ping (latency): The latency between the client’s device and the server should not exceed 150 ms. Higher latency can lead to a deterioration in call quality.
• Packet loss: Acceptable packet loss should not exceed 1-2%. Packet loss affects speech intelligibility and overall call stability.
• Jitter: Jitter (fluctuation in delay) should not exceed 30 ms. Higher jitter values can cause speech distortions and delays.

2. Network Equipment

The equipment used for internet access should support the technologies and features necessary for proper SIP operation.

• Router: The router must support VoIP (Voice over IP) and be capable of configuring QoS (Quality of Service) to prioritize voice traffic.
• NAT and port forwarding: When using NAT (Network Address Translation), port forwarding must be set up for SIP signaling and media traffic. The standard ports for SIP are 5060 (SIP signaling) and ports in the range of 10000-20000 for RTP (Real-Time Transport Protocol).
• STUN/TURN/ICE: For operating behind NAT and bypassing NAT restrictions, it's recommended to use STUN or TURN servers. This ensures proper connection through NAT.
• Updated software: It is important that the client's device (phone, softphone, or IP PBX) has the latest firmware and software version installed to minimize vulnerabilities and ensure stable operation.

3. Support for Necessary Codecs

To transmit voice over the SkyTel network, compatible codecs must be used. The recommended codecs are:

• G.711: A codec with high sound quality that requires approximately 64 kbps per channel. It is used in most cases.
• G.729: A compressed codec requiring 8 kbps but providing acceptable call quality with low internet bandwidth.

4. Firewall and Security Settings

Proper SIP operation requires that the firewall and network security system allow the necessary types of traffic to pass:

• Open ports: Ports for SIP signaling (typically 5060) and RTP traffic (10000-20000) must be open.
• ALG (Application Layer Gateway): Some routers have SIP ALG enabled by default. In most cases, it is recommended to disable it, as it can distort SIP traffic and cause connection issues.
• Encryption: If additional security is required, the TLS protocol can be used for encrypting SIP signaling and SRTP for encrypting media traffic.

5. Redundancy of the Connection

To ensure reliability and minimize downtime, it is recommended to use a backup internet connection (e.g., cellular connection or a second provider). This helps prevent connection loss during outages of the primary internet connection.

6. Quality Control and Monitoring

To ensure stable SIP operation and timely issue resolution, it's important to set up a system for monitoring connection quality and internet channel performance. This may include:

• Network monitoring systems: Using SNMP or specialized software to monitor latency, jitter, bandwidth, and packet loss.
• Voice quality monitoring (MOS): Monitoring call quality using the MOS (Mean Opinion Score) method to assess connection quality.