SIP Trunk Setup and Troubleshooting guide

SIP Trunk Setup and Troubleshooting guide

Introduction

This is a guide to configure your phone systems using Summit's SIP or VoIP account.


Implementation Use Cases

  • Third party SIP Phones
  • SIP compatible PBX
  • SIP compatible Gateways, ATA, etc.

 

Voice Network



Figure 1: Enterprise voice interconnection

Depending on your network setup, we allow voice traffic via Public and Private Network. Please consult with your service manager to know more about your network setup type.


Supported Authentication types

  • Account based authentication: User Agent Client (UAC) is required to send SIP REGISTER message using an account id and password to authenticate  
  • Outbound-calls: User Agent Client (UAC) is NOT required to send SIP REGISTER message however an outgoing INVITE message will be challenged for authentication (407 proxy authentication required)

Note: For Inbound calls and Phone number mapping, it is mandatory to use account based authentication. IP based authentication is unsupported.


Basic Firewall and UAS settings

Note: This configuration is ONLY applicable for UAC on Summit's Private Voice networks (ACL White list)

  • SIP Proxy Address: sip-int.sipvoice.com.au 
  • Supported Transport: TCP and UDP
  • SIP Port: 5060
  • Media Ports: UDP 5060, 10000:60000

Note: This configuration is ONLY applicable for UAC on any public network (ACL White list)

  • SIP Proxy01: sbc1-mel.sipvoice.com.au 
  • SIP Proxy02: sbc1-syd.sipvoice.com.au 
  • Support transport: TCP and UDP
  • SIP Port: TCP 5060
  • Media Ports: UDP 5060, 10000:60000

Dial Pattern

  • Inbound calls: E164 (without plus +) 
  • Outbound calls: e164, FNN, 0011{CC+AC+Number)
  • SIP URI: sip:{account_code}@sip_dns_address

Support codec

  • Priority 1: G711a /PCMA (preferred)
  • Priority 2: G711u/PCMU

Test using soft phone

Via Private Internet

Via Public internet


Having Problem? Follow the basic troubleshooting steps below

Follow the troubleshooting step by step mentioned below in identifying understanding/clarifying the problem area

  1. Check network cable
  2. Ping SIP ATA or Gateway. (Start → command, type: Ping <device_IP_Address>
  3. Check SIP Registration status: Registered or Not registered
  4. Ping SIP Proxy Addresses (Customer → SIP)
  5. Trace route to SIP proxy addresses (DNS) (tracert sip.proxy.address)
  6. Check your Firewall / ACL - IP address, DNS, SIP Ports and RTP port range.
  7. Check sip packet filtering (qos tag)
  8. Check outbound dial string (e164 without plus)
  9. Check media codec (alaw)
  10. Test Inbound call
  11. Test Outbound call


How a SIP session works?


A typical SIP session architecture below  (Invite →  Trying →  Ringing →  200 Ok → ACK → RTP Session → Bye → 200Ok)

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