Skype for Business – Level 400 information

Door buzzers/enterphones, paging systems to Teams or Skype (What is FXO)

Analog trunks and devices are less and less of a factor in many of the projects that I work on. Areas where I still see analog are branch office trunks, faxes, door buzzers/enterphones, paging systems, and ring-down phones that you might see in a parking lot to contact security or a cab company.

To integrate analog trunks or devices with SfB or Teams, you connect them to a gateway. The gateway can come with two types of ports: FXO, and FXS. FXO is an abbreviation for Foreign eXchange Office, and FXS for Foreign eXchange Subscriber.

So, clear as mud and we’re done here, right? If you’re an old school telecom guy, you know there’s a lot of complexity hidden behind that simple jack in the wall. The good news is for nearly all SfB use cases we can boil things down so they’re very simple.

FXO is a port that you plug a telco trunk line into. FXS is a port that you plug your phone or other device into. Mostly. Confusion pops into the picture when you have paging systems or enterphones, which may oddly use the opposite interface than you’re expecting, or give you the option for both.

You can think of FXO and FXS as North and South on a magnet, or male and female, or whatever pairing you’d like. A FXO device plugs into an FXS device, and all is well, like this:

Your phone, which has an FXO jack on it, plugs into the wall, which is an FXS interface.

Your phone, which has an FXO jack on it, plugs into an Analog Gateway such as the AudioCodes MP-118 gateway FXS port, or an FXO/FXS card in a Sangoma for instance.

The AudioCodes MP-114 gateway FXO jack plugs into the wall, which is an FXS interface.

An AudioCodes MP-114 gateway. From the left are power, Ethernet, RS-232 serial console, two FXS ports and two FXO ports.

Great, so why then would a paging system offer both FXO and FXS interfaces? The answer is that there are two different use cases for the paging system.

One use case is a standalone, where a phone plugs directly into the paging system. You pick up the phone, maybe enter some digits to indicate what zone to page, and you talk away. The paging system is acting as a PBX in this scenario.

The second use case is PBX integrated, where the paging system acts as a phone. You dial the extension for the paging system, it rings and then answers, you maybe enter some digits to indicate what zone to page, and you talk away.

These two use cases also apply to things like enterphones or gate/door buzzers. You can have a phone plugged directly into the enterphone, or you have have the enterphone act as an extension on your PBX.

The standalone option is simple, but restricts you to interacting via a single phone. The PBX integrated option is more complex, but allows you to interact via any phone on the PBX.

Caution: “Interact via any phone on the PBX” in the SfB world means that in a global deployment, you could have a prankster user in New York telling jokes over a paging system in Paris. Configure your dial plans appropriately if your paging system doesn’t offer PIN functionality

If you have a choice between using an FXO port or FXS port on a gateway to integrate with an analog device that offers both, I recommend you pick the FXO port. This has the device act as a PBX, which means that there is no ringing when you call it, and call setup is faster. Disconnects are usually quicker too, which is important if the paging system or enterphone is used a lot.

When you configure the device to plug into an FXO port on the gateway, set the gateway to route calls to that number out via the FXO port you’ve connected it to. If the device will be sending calls to the gateway, set the gateway to

You’ll need to use an FXS port on your device to connect to the gateway’s FXO port. If your device has one port that’s switchable between FXO and FXS, read the manual carefully – I’ve seen some that aren’t clear whether they mean FXO mode is “setting this device to FXO” or “setting this device to talk to FXO”. If it’s really unclear, plug a boring analog phone in. If the line is dead, the device is set to act as an FXS device and the port is configured as an FXO interface.

Inside access to external NAT IP services

The corporate office has an environment where there is a separate “guest” network for vendors, visitors, etc. that can use their own devices, to use internet services through Wi-Fi.
Due to the fact some internal services are needed such as joining internal Audio/Video conferences, and access to collaborative services, we had a requirement that access to those services be available through this semi-public network that is “external”, i.e. uses external DNS resolution, but is still “inside” the firewall boundary as shown below.

Access from GuestNet-Diag

To spell it out, we had the following infrastructure:

Internal services only accessible from inside the corporate network and internal devices.
External services accessible from outside the corporate network.
Guest network with external/public DNS resolution.

The requirements are as follows:

Internal services accessible externally if secured with boundary extending services, primarily Microsoft Direct Access, or if explicitly approved, Cisco VPN software.
Skype for Business availability for Guest Wi-Fi to join conferences and collaboration.
SharePoint services (specifically Office Online Server) for collaboration access from Guest Wi-Fi.

Cisco has a good article here: https://www.cisco.com/c/en/us/support/docs/security/asa-5500-x-series-next-generation-firewalls/72273-dns-doctoring-3zones.html to make this process work, the D-NAT methodology was used, but the article can a bit confusing, so I just wanted to explain a couple things to clarify it, and show how the final NAT rules end up looking.

The following are the relevant NAT rules:

Access from GuestNet

For the Match Criteria:
- Source is the Guest network.
- Destination for the Skype for Business services is the “inside” DMZ interface, as in this case, the DMZ is sandwiched, has an internally facing network, and an externally facing network, or wherever the service resides, as some services are on the “services” network.
- IP source is the Guest network object.
- IP destination is the External IP of the service.
For the Action:
- Source is wherever the service resides.
- Destination is the Internal IP of the service.

I hope this helps clarify the Cisco article.

LS Audio/Video Authentication Server Error 19008 – Private Key not found

After happily running for several years, one of the Skype for Business edge servers for one implementation decided it was not going to start its Audio/Video Authentication and Audio/Video Edge service!

Looking at the event viewer, the following two Event IDs were raised: 19008 and 19005. Specifically: 19008: Private key for server certificate not found by the LS A/V Authentication service or the service does not have sufficient permissions to access the certificate.

After verifying the private key permissions are set correctly (NETWORK SERVICE: Read, etc) in the Certificate MMC snap-in, I checked to see what the certs looked like in PowerShell

PS Cert:\LocalMachine\My\> dir .\ 5E670E493E5EBAACC5B26E219ACA8A629F9485D4 | fl 

HasPrivateKey  : True
PrivateKey     :
PublicKey      : System.Security.Cryptography.X509Certificates.PublicKey

Notice that there is no PrivateKey provider defined here, which means the cert broke somehow! Strange, as in this environment there were two Skype for Business edge servers, one worked perfectly, the other did not.

Anyways, the fix was to tear the certs apart, and put them back together as shown in this Merge certificate public and private key with OpenSSL TechNet article.

Specifically

I got the OpenSSL binaries from: https://indy.fulgan.com/SSL

I extracted the keys using the following commands:

openssl pkcs12 -in egdev1.pfx -nocerts -out private_key.pem -nodes
openssl pkcs12 -in egdev1.pfx -nokeys -out public_key.crt

I merged the keys back together using the following command:

openssl pkcs12 -export -in public_key.crt -inkey private_key.pem -out lync_edge_merged.pfx

After certificate import, and applying it to the services, I checked to see what the certs looked like in PowerShell

PS Cert:\LocalMachine\My\> dir .\ 5E670E493E5EBAACC5B26E219ACA8A629F9485D4 | fl 

HasPrivateKey  : True
PrivateKey     : System.Security.Cryptography.RSACryptoServiceProvider
PublicKey      : System.Security.Cryptography.X509Certificates.PublicKey

Notice that there is now a PrivateKey provider defined here, and the two Audio/Video Authentication and Audio/Video Edge services started up just perfectly!

Skype for Business presentation size limits.

I recently had an implementation where very large PowerPoint presentations was needed. When those pptx files were pre-uploaded to the meeting, the following dreaded “allowable file size exceeded” message occurred:

Exceeds File size SfB

It got me interested in finding out what the allowable file sizes are for Skype, and after scouring documentation, I discovered the following:

As of September 2017:

With Office Web Apps 2013, the max file size is 150Mb
With Office Online Server, the max file size is 300Mb

The explicit limits, where applicable, are listed in the table below. However, note that there is a 60-second file download time out that applies to all GetFile operations, and this time out can affect the perceived file size limit. In practice, this time out is rarely hit, since connectivity and bandwidth is typically very good between Office Online and host datacenters. However, hosts should be aware of this limit.

File size limits
Application	Mode	Limit	Notes
Excel Online	View	5MB
Excel Online	Edit	5MB
PowerPoint Online	View	See notes	No limit, but subject to the 60 second time out for file downloads as described above.
PowerPoint Online	Edit	300MB	While the upper limit is 300MB, this is still subject to the overall 60 second time out for file downloads so it is possible that smaller files will hit that timeout.
Word Online	View	See notes	No limit, but subject to the 60 second time out for file downloads as described above.
Word Online	Edit	See notes	The technical limit is 100,000,000 (100 million) characters in the document XML; however, this does not correlate with file size in a meaningful way. For example, a 1000-page document, hundreds of MB in size does not hit this limit. For the vast majority of use-cases, this limit is irrelevant.

The process to configure these max sizes is fairly simple, and is configured in the “Settings_Service.ini” configuration file. The default location for that file is:

C:\Program Files\Microsoft Office Web Apps\PPTConversionService

At the bottom of the file, just add the following entries:

For Office Web Apps 2013, add:

PowerPointEditServerMaxFileSizeBytes=(System.UInt64)153600000
PowerPointServerMediaEmbeddedMaxSize=(System.UInt64)153600000

For Office Online Server, add:

PowerPointEditServerMaxFileSizeBytes=(System.UInt64)307200000
PowerPointServerMediaEmbeddedMaxSize=(System.UInt64)307200000

Once the changes are saved, restart the server service. You may do so in PowerShell with the following command:

Restart-service WACSM

Please note, these max sizes are for the entire meeting, not per attachment, therefore, if your meeting has much larger files, you will have to split them, upload one, go through it, remove it, upload the next. – You can upload several files at a time, but they cannot collectively be larger than the total MB size limits.

Dropped Calls at 10 min with Skype for Business SIP trunk. (& what is SIP ALG?)

The case of the dropped call at the 10 min mark:

We have a SIP trunk provider that due to their implementation of RFC requirements, send down a Re-Invite packet at the 10 minute mark. If you’re just running a single server, there’s no issue, as the FE that gets the packet sends back an ACK packet saying “got it”, and communication continues. When you have multiple servers, it can be an issue, as in this provider’s case, they only send that Re-Invite packet to the first registered IP, and if the outgoing call originates from one of the other servers, they never get that Re-Invite packet, can’t send back an ACK, and the call gets dropped due to timeout.

To mitigate the above mentioned case, an SBC was put in the middle so that the trunk would only have one IP to communicate to, all was fine and dandy, but due to the nature of things changing in an IT environment, a new random call drop issue came up. (Ugh!)

To see what was going on, client and server Skype for Business logs were looked at for the time the drop occurred, they showed a standard “BYE” happening, not an error, so onto looking at the next step, infrastructure.

In order to give some concept of the environment, here is a basic diagram of what it looks like:

Internal Diag

Note that there is NAT’ing going on between the networks, the SBC was configured with the IP 192.168.70.44 (in the syslog figures, it shows up as “Device”) the internal FE servers had NAT addresses of 192.168.70.20, 21, and 22. We’ll get back to that in a bit.

Syslog was enabled on the SBC, perused through those, and this is what I found:

In the process to narrow down what was going on, we found that in the majority of complaints, the calls were getting dropped at the 10 min mark, and since the Re-Invite gets sent at that time, the SIP provider was in the crosshairs, and that was our main focus. After much going back and forth with them, going back and forth with the SBC vender, tearing up the servers, time to start fresh, and look deeper.

In looking at the logs, there are some calls that the re-invites are handled correctly: Here are two long ones, one lasting 13 min, the other almost 30, both with Re-Invite packets, (the second one with three).

Work1-1

Eventually I found some calls that failed, here’s an example of one that wasn’t handled correctly:

Notice the call is only 2 seconds long, but still dropped!

Non0-1

Here’s a screenshot of the Re-Invite hangup problem:

Non3

Here we see the Re-Invite packet coming down at exactly 10 min after the call started (15:08:58), when the internal server answers that with its SDP packet, the address doesn’t get NAT’d, the SBC then tries to send it to the Internal IP, that communication doesn’t occur, and after a period of no communication (14 sec in this instance), the server assumes the call has ended, has the BYE packet sent to drop the call.

So, from looking at all the calls with the syslog viewer, I saw sporadically there were calls failing, and others succeeding, there didn’t seem to be any rhyme or reason to it, except for the failure was always right after an SDP packet coming from the inside server.

This now had me looking at ALG, which in this case, is controlled by the firewall. What is ALG? To put it simply, it is NAT for SIP SDP packets. What is SDP? While SIP deals with creating, modifying, and closing down sessions, SDP deals with the media within those sessions. and what we’re specifically interested in this scenario, is the IP addresses the two endpoints should talk over. Since one of the endpoints is behind a NAT’d device, ALG is the mechanism that changes the data in these packets to have IPs the two devices can talk to!

Thankfully, this firewall makes it easy to capture inbound packets and outbound packets separately in the same session, producing two separate pcap files which can be compared to side by side! Let’s look at packet #24 of a call that does work:

Here is the inbound packet, notice the O and C attributes have the internal IP, as it’s originating from inside the network:

cap1

Once it goes through the firewall, here is the outbound version of that exact same capture, you can see the O and C attributes have been changed to the NAT IP by ALG:

cap2

Here’s an example of a complete failure capture. Note, this was from an “updated” firmware version of the firewall that completely blew up NAT, so the capture was easy to catch, but the concept is the same which is occurring with the sporadic failures above:

Here is the inbound packet #4, notice the O and C attributes have the internal IP, as it’s originating from inside the network:

cap3

Once it goes through the firewall, here is the outbound version of that packet #4, you can see the O and C attributes have NOT been changed to the NAT IP by ALG:

cap4

Let’s take a look at that syslog trace again:

We can see the original invite comes from the Skype FE server (10.50.2.46, which is NAT’d to 192.168.70.22), communication goes to the SBC, and then gets sent up to the SIP provider’s CPE SIP router. Everything works great until one of the SDP packets does not get translated through ALG correctly, when that happens, the SBC does not know where to send the packet, it gets sent to the “Internal” IP, subsequently goes nowhere, and 14 seconds later, the stream is disconnected with a “BYE” packet due to inactivity.

Non1

Why did this problem arise out of the blue? I’m guessing there have been some good firewall firmware versions that have not caused any issues, so there has been some periods of tranquility, with other firmware versions that have “sometimes” caused this issue. The current firewall version installed is 7.1.5. I’ve tried 7.1.8, 8.0, and 8.0.2, they are progressively worse (with 8.0 and higher not working at all), so I went back down to 7.1.5, and am awaiting resolution from the manufacturer.

Another solution is to change the architecture of the network boundaries affected to be Routed instead of NAT’d, that way ALG does not come into the picture. That might not be a feasible solution due to your security or infrastructure constraints, but has solved the issue (temporarily) in this case. It is not a permanent fix, as any SIP conversations going over NAT will continue to be an issue until the vendor resolves it in their firmware.