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Greetings Spiceheads!

I have recently taken a position in a sole IT role that is replacing an MSP. The reasons are vast and the mess is, well – massive. One of the primary complaints and projects I need to tackle, amongst many, is the phone solution. The company that I am working for spent a rather large amount of money on converting from an on-prem PBX to a cloud based VoIP system hosted through. To say the solution has been sub-optimal is being polite, with only 75% of the users having phones configured (let’s leave off the “correct” portion that would normally be included in a configuration statement) and the remainder of the users having paperweights sitting on their desks or being used as art stands.

The provider that was selected was chosen strictly for the “screen pop” (technical user term), AKA the ability to see who is calling via Skype.

While I have gotten all of the phones working and the configuration corrected, there is still an issue of dropped calls, jitter on the line, one direction of the call not going through, etc. We have a mix of Polycom VVX phones, a Cisco SG300-52P (PoE) switch (connected to some outdated Dell PowerConnects), a Cisco ASA 5510, and a 20x20 DIA fiber line. I am replacing the Dell switches with a couple of new Cisco SG300’s, the ASA with a new UTM, and the 20x20 with a 100x100 DIA fiber line, but I’m still not sold that replacing the outdated equipment and beefing the internet pipe is going to be enough to correct the phone issue.

To add another wrinkle, we are looking at going with a Revolabs FLX2 conferencing phone so that we can move it between floors and conference rooms with ease, but our current provider is stating that anything that we attempt to use outside of a very narrow list of phones that they support will essentially void any support from them. They will still bill us for support we just won’t be able to request anything from them……Never mind.

I have been told that we need to keep the phone system that we are using, which basically means they want to stick with VoIP, but that if the integration with Skype needs to go in order to drop our overall bill/switch providers then it will have to go.

I’m looking for thoughts and options, outside of what I have listed, for improving the call quality for my end users.

Thanks!

5 Spice ups
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I don’t think your equipment is your issue…I’d be looking at the company you’re using as the issue, or the connection between your ISP and them

3 Spice ups
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I would check the call transmission codecs first, also make sure your network supports QoS and it is configured on the switches and your firewall.

We just switched ShoreTel providers as our previous company was taking 6+ months to resolve call quality issues that the new company fixed in less than 2 weeks.

The issue turned out to be our codecs were not optimized for our different connection mediums (T1 PRI vs Ethernet VPN).

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Equipment was unlikely to ever have been the issue. Sticking with VoIP is the only sane choice, this isn’t the 1990s. ALso, Skype is VoIP.

1 Spice up
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I’d pick a sample set of devices in one location and study the related network traffic from the phones through your network to the edge at the ISP. The point being to identify what each call is using for bandwidth and whether the edge of your network to the ISP is handling it. This is where your QOS / traffic shapping or policing is going to happen.

1 Spice up
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Looking at your equipment list my first inclination is to tell you to check the settings on the ASA 5510. Make absolutely certain that the VOIP bypasses any firewall or stateful packet inspection as that will queue up data packets and kill VOIP.

3 Spice ups
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Good luck - sounds like a quagmire. If you have any questions on FLX2 or our other conference phones, give our support people a call/chat/email; they’re very knowledgeable and helpful.

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You can do things like test calls from home, too. Bypass the network completely to try to isolate where problems might exist.

1 Spice up
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Hey Steve- if you’re looking at switching providers, I just wanted to make sure you knew about Intermedia’s Hosted PBX. You can add enterprise-grade calling features, improve reliability, and enable mobile users—while cutting your phone bills by over 50%.

Linking to more info here, but feel free to reach out with questions: PBX Phone Systems for Enterprise & Small Business | Intermedia

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Wireshark, get a packet capture from somewhere that you can see both sides of the conversation, your gateway might be a good choice as you will see the convo initiated on the internal interface and replies on the external interface. If your gateway does not have the function to do this the mirror the gateway port on the switch and capture the traffic from a laptop.

Capture packets for a rolling window say 3 days, each time someone reports and issue save the file an examine for issues.

I have had issues before where the ISP providing the SIP service was responding in a non-standard way (prompting for codec to be used instead of accepting the default) to the phone system and the default behavior of the phone system was to resend the request therefore dropping the call after too many attempts.

If you don’t find any issues when monitoring from the gateway then it is likely an up stream issue and you will need to start contacting providers.

2 Spice ups
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As others have said it sounds like the network may be where the trouble lies.

Wire shark can be your friend here. Pick out some trouble spots and find out exactly what’s going on with a capture. While it can be a tedious endeavor you should be able to see the trees from the forest on what’s going on.

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Mark - thanks for the reminder to check the settings! It’s not the first issue that I have had with this firewall. This is one of my top replacement projects!

Thanks!

1 Spice up
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Scott - Thanks for the recommendation. Took a phone home last night and had no issues connecting up, making or receiving calls. Call quality was much better from home as well.

2 Spice ups
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Who is the provider?

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Definitely make sure the provider is serving up packet prioritization and that it’s configured properly for the traffic from your edge device. Also, I think it was mentioned before but make sure your ASA is passing VoIP traffic without SPI or anything like that. Jitter will make it useless.