The below steps provide guidance on integrating an AudioCodes MediaPack ATAs with LoopUp to provide service to Fax and Analogue devices.
Step 1: Prerequisites
Please note that all steps below assume that the device is online and operational on your local network with the following configuration in place:
- Default credentials have been changed
- Static local IP address
- Accessible from the internet via one of the below methods
- Direct public IP address allocation
- NAT/PAT
- If TLS is to be configured customer provided public certificate will be required
Step 2: LoopUp SIP Trunk Provisioning
- Navigate to the SIP Trunking Setup article and follow the process to deploy a SIP Trunk with LoopUp
Step 3: Firewall Rules
The below rules will need to be in place to allow the required signalling and media traffic.
Source & Destination
Depending on the region selected, utilise the below as the Source LoopUp Endpoint 1 & 2 for your firewall rules:
- AMER
- usdc1-edgesbc.loopup.com - 176.74.7.53
- usdc2-edgesbc.loopup.com - 176.74.3.253
- EMEA
- ukdc1-edgesbc.loopup.com - 176.74.5.5
- ukdc2-edgesbc.loopup.com - 185.66.35.35
- APAC
- ausdc1-edgesbc.loopup.com - 176.74.1.4
- hkdc1-edgesbc.loopup.com - 185.66.34.4
The Destination for your firewall rules will be the ATA Public IP.
Signalling
- Source Port: Any
- Destination Port: UDP/TCP:5060 or TLS:5061
Media
- Source Port: Any
- Destination Port: UDP 7000 - 65535
Step 4: Device Configuration
Security Settings
VoIP > Security > Firewall Settings
- No Firewall rules configured
VoIP > Security > General Security Settings
- Enable IP Security: Disabled
If using TLS, then also set the following
- TLS Version: TLSv1.2
- TLS Mutual Authentication: Enable
- TLS Client Verify Server Certificate: Enable
- TLS Remote Subject Name: [External FQDN of Device]
Media Settings
VoIP > Media > RTP/RTCP Settings
- RTP Base UDP Port: 10000
VoIP > Media > Media Security
- Media Security: Enable
- Media Security Behaviour: Mandatory
- SRTP Tunnelling Authentication for RTP: Enable
- SRTP Tunnelling Authentication for RTCP: Enable
- SRTP Offered Suites:
- CIPHER SUITES AES CM 128 HMAC SHA1 80
- CIPHER SUITES AES CM 128 HMAC SHA1 32
- CIPHER SUITES ARIA CM 128 HMAC SHA1 80
- CIPHER SUITES ARIA CM 192 HMAC SHA1 80
Control Network Settings
VoIP > Control Network > Proxy Sets Table
Depending on the protocol and region selected, fill out the Primary SIP Server and Failover SIP Server as follows:
For UDP or TCP replace port below with 5060, for TLS use 5061
- Proxy Address
- AMER
- usdc1-edgesbc.loopup.com:port
- usdc2-edgesbc.loopup.com:port
- EMEA
- ukdc1-edgesbc.loopup.com:port
- ukdc2-edgesbc.loopup.com:port
- APAC
- ausdc1-edgesbc.loopup.com:port
- hkdc1-edgesbc.loopup.com:port
- AMER
- Enable Proxy Keep Alive: Using Options
SIP Definitions
VoIP > SIP Definitions > General Parameters
- NAT IP Address: [Public IP for the site]
- Enable Early Media: Enable
If using TLS, then also set the following
- SIP Transport Type: TLS
- SIP TLS Local Port: 5061
- Enable SIPS: Enable
VoIP > SIP Definitions > Proxy & Registration
- Use Default Proxy: Yes
- Always Use Proxy: Enable
GW and IP to IP Settings
VoIP > GW and IP to IP > Hunt Group > Endpoint Phone Number
- [Select Channel]
- Phone Number [Device Number]
VoIP > GW and IP to IP > Manipulations > Dest Number Tel -> IP
Rule
- Index: 0
- Destination Prefix: 0
- Source Prefix: *
- Source Trunk Group: -1
- Destination IP Group: -1
Action
- Index: 0
- Stripped Digits From Left: 0
- Stripped Digits From Right: 0
- Number of Digits to Leave: 255
- Prefix to Add: +44
- Suffix to Add: Blank
VoIP > GW and IP to IP > Routing > Tel to IP Routing
- Src. Hunt Group ID: *
- Dest. Phone Prefix: *
- Source Phone Prefix: *
- Dest. IP Address
- Choose from the below:
- AMER
- usdc1-edgesbc.loopup.com - 176.74.7.53
- usdc2-edgesbc.loopup.com - 176.74.3.253
- EMEA
- ukdc1-edgesbc.loopup.com - 176.74.5.5
- ukdc2-edgesbc.loopup.com - 185.66.35.35
- APAC
- ausdc1-edgesbc.loopup.com - 176.74.1.4
- hkdc1-edgesbc.loopup.com - 185.66.34.4
- AMER
- Choose from the below:
- Port: UDP/TCP 5060 or TLS 5061
- Dest. IP Group ID: -1
- IP Profile ID: 0
- Cost Group ID: None
VoIP > GW and IP to IP > DTMF and Supplementary > DTMF & Dialling
- Max Digits In Phone Num: 15
Troubleshooting
Admin Guide
https://www.audiocodes.com/media/13280/mp-11x-and-mp-124-sip-users-manual-ver-66.pdf
Troubleshooting Guide
https://www.audiocodes.com/media/10408/ltrt-57601-cpe-sip-troubleshooting-guide.pdf