Configuring Arkan Integrated Solutions SIP Trunk on a PBX System

📌 Prerequisites

Before you begin:

Ensure you have an active Arkan Solutions SIP service plan.
At least one active DID (Direct Inward Dialing) number tied to your subscription.
Firewall/NAT rules allowing SIP signaling and media traffic to/from Arkan proxy and media IPs.

81.208.160.76
81.208.162.84
144.24.220.42
84.8.105.58
84.8.96.242
84.8.123.48
84.235.250.21
193.123.88.110
193.123.69.207

For Saudi Arabia services, the customer's IP must be within Saudi Arabia.
For UAE services, the customer's IP must be within the UAE.

📶 SIP Signaling Settings

Arkan Proxy Servers

Proxy Server	IP / FQDN	Port	Protocol	TLS Notes
Primary	proxy001.arkanet.net / proxy001ruh.arkanet.net / proxy001dxb.arkanet.net	5060	TCP/UDP	Standard SIP (unencrypted)
Primary	proxy001.arkanet.net / proxy001ruh.arkanet.net / proxy001dxb.arkanet.net	5061	TLS	Use TLS v1.2 only (provider requires TLS 1.2 for signaling encryption)

TLS Version Requirement: Only TLS 1.2 is permitted for encrypted SIP signaling (SIP over TLS).
Make sure your PBX/SBC supports TLS 1.2 on port 5061 (custom port is allowed if requested) and that certificate validation is properly configured if applicable.

🎧 Media (Audio) Configuration

Arkan uses the same media server IPs listed under signaling or will negotiate via SDP in the call — that means media (RTP/SRTP) will be determined dynamically during the SIP session. The PBX/SBC should support encrypted media offers. However, enforcing non encrypted media on the trunk must be requested from Arkan team subject to regulations of the country of the service purchased.

Media Transport Modes

You should configure your PBX/SBC to accept and offer SRTP options to ensure interoperability regardless of the endpoint’s media encryption support:

Media Mode	Description	Recommended Setting
Encrypted Media (SRTP)	Secure Real-Time Transport Protocol must be used if both sides support it.	Offer SRTP first (Preferred)
Unencrypted Media (RTP)	Fallback to legacy RTP if SRTP negotiation fails.	Accept RTP

Recommended Configuration:

Accept both SRTP offers on incoming and outgoing calls.
Prefer SRTP when available (ensure your PBX/SBC supports this).
Do not reject unencrypted RTP unless your environment requires strict media encryption.

TB: Many PBX/SBC platforms call this setting “SRTP optional/allow SRTP or RTP” — configure that mode rather than “SRTP only” to allow fallback.

📡 Media Port Ranges

You must allow the media (RTP/SRTP) port range on your firewall to/from Arkan networks.
Typical ranges used by SIP providers include UDP ports 10000–60000 (exact range depends on your PBX).
Ensure these are open and accessible for two-way audio.

🎙️ Codec Support & Priorities

To maximize interoperability and call quality, set your codec preferences as follows:

Preferred Codec Order

G.711 A-law / µ-law (highest priority)
- Best compatibility and PSTN interoperability.
Opus
- Wideband codec (if supported by both sides).
G.729
- Low bandwidth (optional based on licensing).
G.722
- Wideband codec (good quality).

Supported Codecs

Preferred / Recommended:
- G.711 A-law, G.711 µ-law
Also Supported:
- Opus, G.729, G.722

Ensure your trunk configuration uses G.711 as the primary codec, with the others enabled below it in priority.

🔐 Encryption Summary

SIP Signaling: TLS over port 5061 using TLS 1.2 only.
Media Encryption: SRTP preferred; allow RTP fallback for compatibility.
Codec Priority:
- Highest preference: G.711 a/µ-law
- Supported: Opus, G.729, G.722.

✅ Example SIP Trunk Configuration Checklist

✔ Transport Protocol: TLS 1.2 on port 5061
✔ SIP Proxy IPs Whitelisted
✔ Media Ports (UDP 10000–60000) Allowed
✔ Media Mode: SRTP (preferred) + RTP allowed
✔ Codec Order Set (G.711 → Opus → G.729 → G.722)
✔ NAT/Firewall Rules for both SIP & RTP/SRTP