View source for Network:Firewall:Configuration

{{Blackboxwarning|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}}
{{Warning|SIP ALG: Because many implementations of SIP ALG are not working correctly (bug or incompatibility with some phones), we have configured the VoIP platform to handle NAT all the time. SO PLEASE DO NOT USE "SIP ALG", it is not needed}}
{{Warning|STUN / TURN: Because STUN is not working in call cases (like symmetrical NAT), we have configured the VoIP platform to handle NAT all the time. SO PLEASE DO NOT USE "STUN" nor "TURN", it is not needed}}
{{Notice|If you have trouble with your firewall, we recommend to start with a simple configuration allowing all streams from LAN to WAN, then if it is working, you can add filtering rules}}


=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: <font color="green">[[NOC:Hardware:Sonicwall|SonicWall]] / [[Tools:iptables|Iptables]] are OK by default =>  30s</font>
**ex: <font color="green">[[VoIP:Hardware:Thomson Tips|Thomson / Technicolor]] are OK by default => 124s</font>
*'''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

{{Warning|On Sonicwalls, if you have a MTU 1492 on the WAN, please set "Do not send ICMP Fragmentation Needed for outbound packets over the Interface MTU", otherwise it will break calls for snom phones that will display "Network Failure", cf screenshot below}}


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

{{Warning|Please do not set up any NAT detection, no STUN, no TURN, no SIP ALG, no externip !}}

*'''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==


{{Blackboxwarning|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}}


{{Notice|By default a firewall performing NAT should be stateful and might allow any PC or phone to connect from the LAN to the internet, if you kept the default mode, you can skip this section. If by default you decided to block everything for outgoing traffic (i.e: by default PC and Phone on the LAN cannot connect to the internet) except if it is explicitly allowed, then use this section where all needed traffic is listed}}

===Phone / VPBX / Connect / Firewall rules===

{{Warning|1=Some customer might complain that the 212.147.44.0/22 is too big, in this case, please use 212.147.47.208/28 instead in the following rules (which is the current Production platform ). This /22 contains all our VoIP infrastructures for Lab + PreProd + Prod, that is why it is so big ( contains 1024 IPs ).<br/> If you wish to know which IP is being used right now by your phones for SIP/SIPS and RTP/SRTP, you need to do a DNS resolution of your SIP domain. It should be right now 212.147.47.217 (for VPBX) or 212.147.47.218 (for ConnectPBX). WARNING, this IPs  might change over time, this is why we gave you a wider range. If in the future we change this IP and you setup too narrow rules, your phones won't be able to connect anymore}}
{{Warning|1=If you own Yealink phones, you will need to allow them to perform NTP queries towards pool.ntp.org which has dynamic IPs ( so allow NTP towards all the internet ), otherwise it won't boot. As a workaround, you could try to redirect NTP queries and have them answered}}
{{Notice|1=@VTX: If you do alter the list below, please inform NOC team to have the ACLs on the LNS and if needed the STS of VTXBox updated}}

*'''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'' ( <font color="red"><b>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</b></font> )
*'''HTTP/HTTPs''' for phone auto-configuration
**'''IP range''': ''secure-provisioning.snom.com + rcs.aastra.com + prov.gigaset.net + profile.gigaset.net + suota.gigaset.com + 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'' + ( <font color="red"><b>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 </b></font>)
**'''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



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===

{{Warning|We have enabled Media Bypass to have the Secured RTP traffic going directly from the Teams Phone to VTX SBC without going via the Microsoft Teams Cloud to reduce call latency. So if you have restrictions on your firewall, you also need to allow VTX Specific IPs}}
{{Warning|Microsoft did enlarged IP Ranges used by Microsoft in June 2022. Thanks to recheck your Firewall settings and adapt ranges if needed}}

*'''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 Ranges: 52.112.0.0/14 + 52.120.0.0/14
**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 Ranges: 52.112.0.0/14 (IP addresses from 52.112.0.1 to 52.115.255.254). + 52.120.0.0/14 (IP addresses from 52.120.0.1 to 52.123.255.254).
**UDP/SRTP Source Port Range : 3478-3481 and 49152 – 53247

*'''RTP Media Bypass Firewalling''': This is used for Media Bypass
**RTP IP Ranges: 52.112.0.0/14 (IP addresses from 52.112.0.1 to 52.115.255.254). + 52.120.0.0/14 (IP addresses from 52.120.0.1 to 52.123.255.254).
**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===

{{:FAQ:VoIP:GeoIP_Filtering}}

===Other Network Restrictions===
{{Warning|Some firewalls or remote ISP might also restrict VoIP, you can use the following tests to verify connectivity}}
{{Notice|If you have some connectivity problem, please perform the tests below, check the trace on your firewall and inform our support from which IP did you test it and at which exact time}}

*''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==

{{Blackboxwarning|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 !}}
{{Notice|If you have trouble with your firewall, we recommend to start with a simple configuration allowing all streams from LAN to WAN, then if it is working, you can add filtering rules}}


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

*'''Modifications to perform on the Sonicwall'''
**'''VoIP''' / '''Settings''' / '''Enable Consistant NAT''' => [http://help.mysonicwall.com/sw/eng/1531/ui2/13000/Firewall/VoIP.htm Sonicwall Doc Helper]
**On the WAN interface, please set '''"Do not send ICMP Fragmentation Needed for outbound packets over the Interface MTU"'''

<table>
<tr><td>
[[Image:SonicWall-VoiP-Config.png|thumb|600px|left|Sonicwall - Disable SIP ALG]]
</td><td>
[[Image:SonicWall-Network-Config.png|thumb|600px|left|Sonicwall - Suppress ICMP Fragmentation Needed message generation if Sonicwall is doing PPPoE with MTU 1492]]
</td></tr>
</table>
<br>

==Watchguard M400 / Snom==

{{Warning|On Watchguard, if used together with Snom, possible '''call cut''' and '''audio problemes''' - Watchguard should get updated to Firmware: 12.1.3.B563398 (Fireware OS)}}

==Connection UPC / check Modem==

{{Notice|we have some customers with an UPC connection using VTX VoIP-Services and having problems which could get solved by get a different type of UPC Modem (@VTX: cf example in G2K ticket 1622489)}}