Network:Firewall:Configuration

Please make sure your firewall is running the last stable version of his firmware and is not End Of Life or End Of Support because we coped with many bugs on Zyxel / Sophos / Sonicwall / ... firewalls running old versions

Configuration

Firewall NAT and QOS configuration

• NAT Timeout: Set the NAT/firewall UDP timeout to a minimum of 30s for SIP/UDP and 600s for SIP/TCP
  • ex: SonicWall / Iptables are OK by default => 30s
  • ex: Thomson / Technicolor are OK by default => 124s
• QOS: You need to reserve 100kbps per concurrent call and 100 kbps per 10 BLF NOTIFY message (LED blinking in the same time on the phones)
• SIP ALG: To disable ( cf warning above)
• MTU: Please make sure you are using the good MTU value on the LAN and WAN interface of your firewall
  • 1500 on the LAN
  • 1500 on the WAN if Firewall is only doing routing
  • 1492 on the WAN if Firewall is doing PPPoE

Network and NAT setup : STUN / TURN / externip / SIP ALG

• Problem 1: The usage or STUN or TURN or SIP ALG is useful to perform some peer to peer VoIP communication, but these protocols are not working in 100% of cases ( i.e: when using symmetrical NAT (almost all firewall now), or having a firewall that do not support hairpin or do not allow LAN to LAN communication ( like in Hotel Rooms ) )
• Problem 2: The other problem is that some implementations of these protocols are buggy on some equipments ( phone or PBX or firewall )
• Conclusion: Since these protocols are not working in all cases and are sometimes buggy, the VTX VoIP platform is always handling NAT detection and all VoIP stream goes via the VTX VoIP platform. Consequently, no need to enable these protocols

Access List

You do not need any incoming firewall rule or NAT rules for your phones to work correctly. You need your firewall to be stateful and allow incoming traffic that have been triggered by an outgoing request. Please follow 1st chapter. ex: phone is REGISTERING on the platform and maintain it each 30s to allow incoming calls to work because the VoIP platform will use this same connection to send incoming calls

Phone / VPBX / Connect / Firewall rules

• SIP signaling that allows your phone to call out and to receive calls
  • IP range: 212.147.44.0/22 (from 212.147.44.0 to 212.147.47.255).
  • Port range: UDP/5060 + TCP/5060 + TCP/5061 (for SIP/TLS)
• RTP and RTCP packets that transport the voice and quality call data
  • IP range: 212.147.44.0/22 (from 212.147.44.0 to 212.147.47.255)
  • Port range: all UDP ports ( WARNING : If you really need a range, use 1024->65535. DO NOT TRY TO GUESS THE PORT RANGE WE ARE USING, IT IS CHANGING OVER TIME WITH CAPACITY INCREASE )
• HTTP/HTTPs for phone auto-configuration
  • IP range: secure-provisioning.snom.com + rcs.aastra.com + prov.gigaset.net + profile.gigaset.net + rps.yealink.com + 212.40.12.0/24 + 212.147.44.0/22
  • Port range: TCP/80 + TCP/443
• LDAP/LDAPS for Centralized Kiosk Directory
  • IP range: 212.40.12.0/24
  • Port range: TCP/389 + TCP/636
• DNS/NTP for DNS queries and NTP time updated
  • IP range: rs0[1-4].vtx.ch + ( WARNING: Allow "*" in case you own Yealink phones because the Yealink won't boot if there is no answer towards pool.ntp.org which has dynamic IPs )
  • Port: UDP/53 + TCP/53 + UDP/123
• Syslog: Used for VTX Support to help debug any problem with auto provisioned phones
  • IP range: 212.147.99.16/28
  • Port range: UDP/514

NB: Regarding RTP configuration, we do not recommend to set a UDP port range restriction because we can add ranges without notice if needed

Here is an overview of an Tools:iptables configuration

### Allow SIP Signalisation ###
-A FORWARD -p udp -m multiport --dports 5060 -d 212.147.44.0/22 -j ACCEPT
-A FORWARD -p tcp -m multiport --dports 5060,5061 -d 212.147.44.0/22 -j ACCEPT
# Allow all TCP ports for MS Skype for Business
-A FORWARD -p tcp -d 212.147.44.0/22 -j ACCEPT
### Allow RTP (voice) ####
-A FORWARD -p udp -d 212.147.44.0/22 -j ACCEPT
### Allow HTTP/HTTPS ###
-A FORWARD -p tcp -m multiport --dports 80,443 -d secure-provisioning.snom.com,rcs.aastra.com,prov.gigaset.net,profile.gigaset.net,rps.yealink.com,212.40.12.0/24,212.147.44.0/22 -j ACCEPT
### Allow LDAP/LDAPS ###
-A FORWARD -p tcp -m multiport --dports 389,636 -d 212.40.12.0/24 -j ACCEPT
### Allow DNS/NTP ###
-A FORWARD -p udp -m multiport --dports 53,123 -d rs01.vtx.ch,rs02.vtx.ch,rs03.vtx.ch,rs04.vtx.ch -j ACCEPT
-A FORWARD -p tcp -m multiport --dports 53 -d rs01.vtx.ch,rs02.vtx.ch,rs03.vtx.ch,rs04.vtx.ch -j ACCEPT
### Allow all NTP servers for Yealink phones that needs NTP towards pool.ntp.org to work at boot process otherwise they won't boot, and since pool.ntp.org is dynamic, all NTP traffic needs to be allowed ###
-A FORWARD -p udp -m multiport --dports 123 -j ACCEPT
### Allow Syslog ###
-A FORWARD -p tcp -m multiport --dports 514 -d 212.147.99.16/28 -j ACCEPT
-A FORWARD -p udp -m multiport --dports 514 -d 212.147.99.16/28 -j ACCEPT

Teams Connect/Virtual Firewall Rules

• Information: The list of List of IPs for Office 365 services for Skype for Business and Microsoft Teams available for Switzerland is available at https://docs.microsoft.com/en-us/microsoftteams/direct-routing-plan#sip-signaling-fqdns-and-firewall-ports

• SIP Signaling: Here are the FQDN to use to define the destination of the trunk
  • sip.pstnhub.microsoft.com – Global FQDN – must be tried first.
  • sip2.pstnhub.microsoft.com – Secondary FQDN – geographically maps to the second priority region.
  • sip3.pstnhub.microsoft.com – Tertiary FQDN – geographically maps to the third priority region.

• SIP/TLS Firewalling: The SIP/TLS flows could come from one of these IPs and TCP Range
  • IP Range: 52.114.148.0 + 52.114.132.46 + 52.114.75.24 + 52.114.76.76 + 52.114.7.24 + 52.114.14.70
  • SIP/TLS Source Port Range: 1024 – 65535

• RTP Transport Relay Firewalling: The media traffic flows to and from a separate service in the Microsoft Cloud. The IP range for Media traffic:
  • RTP IP Range: 52.112.0.0 /14 (IP addresses from 52.112.0.1 to 52.115.255.254).
  • UDP/SRTP Source Port Range : 49152 – 53247

• RTP Media Bypass Firewalling: This is used for Media Bypass
  • RTP IP Range: 52.112.0.0 /14 (IP addresses from 52.112.0.1 to 52.115.255.254).
  • UDP/SRTP Source Port Range : 50000 – 59999

• VTX RTP Media Bypass Range: This is used by VTX SBCs
  • RTP IP Range: 212.147.44.0/22 (IP addresses from 212.147.44.0 to 212.147.47.255).
  • UDP/SRTP Source Port Range : 10000 - 39999

VTX GeoIP Restriction

Here are the different GeoIP restrictions on the Telephony platform

• Physical phone auto provisioning: is only working from Switzerland and France ( other location could be added on request )
• VTX Softphones : Could be used anywhere in the world ( but you still need to allow outgoing calls if calling from abroad, cf below )
• Register and Incoming Calls: Open to the whole world
• Outgoing Calls: Billed outgoing calls are only allowed from Switzerland by default ( This setting can be changed from customer Admin Kiosk interface to open it to Europe or World or some specific IPs) (@VTX: Please refer to Network:Firewall:Configuration#GeoIP_Filtering for more details)

• These settings can be found under "My Services - Telephony - Call restrictions - IP Filtering"
• The "IP Filtering" tab is only visible if you access the kiosk from a swiss IP address (otherwise this tab is hidden)
• A customer can't do any adjustments if he is already abroad. In such case these changes would have to be done by someone who has access from an IP in Switzerland, or uses TeamViewer / Remote access / VPN / proxy server for the modification.

Other Network Restrictions

• Ping the VoIP platform which is opened to ping from the whole world

ping s1.12345.bus.ipvoip.ch
PING vtx.res.ipvoip.ch (212.147.47.217) 56(84) bytes of data.
64 bytes from fix.47.147.212.vtx.ch (212.147.47.217): icmp_seq=1 ttl=53 time=7.65 ms
64 bytes from fix.47.147.212.vtx.ch (212.147.47.217): icmp_seq=2 ttl=53 time=8.32 ms
64 bytes from fix.47.147.212.vtx.ch (212.147.47.217): icmp_seq=3 ttl=53 time=8.14 ms
--- s1.12345.bus.ipvoip.ch ping statistics ---
3 packets transmitted, 3 received, 0% packet loss, time 2002ms
rtt min/avg/max/mdev = 7.659/8.041/8.324/0.289 ms

• SIP is also allowed to work on TCP, so you can use telnet to verify your connectivity towards the platform (you need to see "connected" message)

telnet s1.12345.bus.ipvoip.ch 5060
Trying 212.147.47.217...
Connected to s1.12345.bus.ipvoip.ch.
Escape character is '^]'.

Vendor Specific Setup

Sonicwall VoiP Configuration

Please do not forget to enable Consistant NAT on the Sonicwall, we have noticed bugs on Sonicwall with Downstream audio problem after some time when disabled !

You can find here the VoiP parameters for the sonicwall validate by VTX.

• Modifications to perform on the Sonicwall
  • VoIP / Settings / Enable Consistant NAT => Sonicwall Doc Helper
  • On the WAN interface, please set "Do not send ICMP Fragmentation Needed for outbound packets over the Interface MTU"

Sonicwall - Disable SIP ALG

Sonicwall - Suppress ICMP Fragmentation Needed message generation if Sonicwall is doing PPPoE with MTU 1492