Wanting to eat our own dog food and wash it down with a big tumbler of kool-aid, my team recently held a meeting of nationally dispersed attendees and used the GoToMeeting VoIP features. I'm not kidding when I say I haven't heard that much reverb, distortion and echo since the last time I listened to "The Piper at the Gates of Dawn." As our first attempt, we spent quite a bit of time complaining to each other about the sound quality and asking each other to place our devices on mute. By the way, did you know that the default setting in the GoToMeeting preferences is to always save chat logs? The following is an extract of the recorded GoToMeeting chat that occurred. Names have been changed to protect the innocent.

B (to All - Entire Audience): the voice quality is terrible
K (to All - Entire Audience): you have a lot of reverb and I can't understand what you're saying
R (to All - Entire Audience): can someone mute their mic
R (to All - Entire Audience): massive echo
V (to All - Entire Audience): Click on the green mic icon and you can mute it
B (to All - Entire Audience): cant understand a word this other speaker is saying

I don't have specific stats yet as to how many people were on "regular" phones vs. using computer mics, but judging on the icons in the attendee list it was a nice enough mix of what one would probably reasonably encounter in this scenario at other companies.

Now, I'm used to using our stuff before it's released and dealing with the intricacies of things that don't quite work yet, but in this case it's not the application - it's the settings. In this case it's just a matter of understanding that one-size does not always fit all. Turns out if we had done a little pre-meeting training and all made some quick and easy settings adjustments to our individual GoToMeeting installations, we could have had a much more satisfactory experience.

When in doubt, read the friendly manual

If you've experienced similar issues or haven't tried this feature yet, there's a good article in the GoToMeeting online help to mitigate this -

GoToMeeting VoIP Audio Best Practices

First, what device are you planning to use? Check out the chart in the link above for recommendations. A USB headset connected to your computer will offer the best quality experience, while using your laptop's built-in microphone and speakers will give you a poor experience, especially if your mic is picking up what's coming out of your speakers - echo city.

And here's an additional excerpt that may help:
VoIP Audio Setup - PC

1. Right-click the GTM icon in the PC system tray and select Preferences.
2. Select Audio.
   Microphone Setup - It is recommended that you test your microphone. To test, select your microphone device from the drop-down menu and speak into your microphone; if it is connected correctly, the sound meter will light up green. If the green meter does not light up, select another device listed in the drop-down menu and repeat this test.
   Speakers Setup - It is recommended that you test your speakers. To test, select your speaker device from the drop-down menu and click Play Sound; if connected correctly, you will see the sound meter light up green and hear a soundtrack through your speaker device. If you do not hear sound after clicking Play Sound, select another device listed in the drop-down menu and repeat this test.
   Advanced - GoToMeeting automatically adjusts audio levels. We recommend you keep this checked. If you uncheck this selection, you must manually configure your audio settings through Windows Sounds and Audio Devices. If your attendees can't understand you because your voice is distorted, try unchecking "Microphone boost."
3. Click OK.
   
   

I strongly recommend that you read the rest of this article, consider using a USB headset and adjust your microphone and speakers settings before joining your next GoToMeeting VoIP call for a much better experience.

If all else fails, the meeting organizer can mute/unmute all participants by selecting *5

Have a happy meeting!

Ok, I think this would be an interesting one for the community. Audio (voice & sound) content is becoming more important to customers - day-to-day applications that require clear audio (e.g., listening to a voicemail attached in your email), VoIP/softphone calls, dictation applications, etc. If you have applications or use cases that require audio (voice or sound) over XenApp, please read on...
Take a listen to the recorded audio clip (no video) below - the clip was recorded over XenApp to an ICA client. As you listen to the clip, think about whether the audio quality is, in your opinion, good enough for you day-to-day use? After listening to the audio clip, please vote in our poll below. By the way, it's recommended that you watch/listen to the clip from a locally installed browser.

Oh, and if you cannot see the voting buttons (green checks).... you'll need to login to vote. And if you don't have a login, you can easily create one here.

What's your impression of the audio quality in the sample audio clip?	Choose
Poor – if it's my only option, then I cannot use it with my voice/audio application
Bearable – I can use it to communicate with others, but only if I have to
Acceptable – I can use this on a daily basis
Great – When can I have this?

If you have questions, please click on my name and feel free to contact me!

Recently, I blogged about three ways of delivering telephony functions to Citrix Presentation Server users. A reader commented: "I currently use the Avaya Softphone via Citrix for Remote Call Center agents, allowing complete in-the-office phone features that allow us to not rent office space." You can learn more about publishing the Avaya IP Softphone in the following document on Avaya's web site: http://support.avaya.com/elmodocs2/ip_agent/R7.0/125772.pdf

Derek Thorslund
Product Strategist, Multimedia Virtualization

One of the most viewed Citrix blog posts of all time is Orestes Melgarejo's post, "To VoIP or not to VoIP". A lot of customers have expressed interest in delivering telephony functions to their users via Presentation Server. The purpose of this blog post is to highlight that there are at least three ways you can already deliver telephony functions with Presentation Server 4.5 today. I'll get into more detail on each of these currently available solutions in subsequent posts, but let's start with a brief summary:

1. Published softphone in "control mode".

There are softphones available today from vendors such as Avaya and Cisco that are Microsoft Terminal Services compatible and can be published on Presentation Server to control a physical telephone set. Likewise, Microsoft's Office Communicator client, a Unified Communications tool, can be used in this way. In control mode, you use the softphone application to make calls, redirect calls to any telephone, establish audio conferences, control telephony features, and so on.

2. Softphone application streamed to the user's Windows PC.

The application streaming feature of Presentation Server 4.5 Enterprise Edition and Platinum Edition can be used to stream compatible softphone applications to client devices. The softphone application then runs within a protected isolation environment. The softphone is never actually installed on the user's PC. This approach to softphone delivery centralizes management and reduces the risk of operating system instability from application conflicts.

At iForum in Las Vegas last year, Citrix, HP and Cisco jointly demonstrated softphone streaming to a Windows XPe thin client. Enabling telephony is one of the common functions of an embedded operating system. We're still accepting requests from customers who would like to field test this solution. Amazingly, it was all done with generally available product releases.

3. EasyCall.

EasyCall, powered by the Citrix Communications Gateway, is a feature of Presentation Server Platinum Edition. EasyCall provides a remote telecommunications capability without the need to purchase and support softphones. With its click-to-call function, users can click on any telephone number that appears on their screen, whether in a directory or any other application, and EasyCall will initiate a call to that number. EasyCall does this by calling the user first and then completing the call to the number on the user's screen. The user can specify whether they are using their office phone, home phone, mobile phone, softphone or whatever. Since all calls placed through EasyCall actually originate from the company's telephone system, long distance charges are reduced.

In addition to the three solutions summarized above, the Citrix Access Gateway deserves a mention in this context. The Access Gateway is capable of tunneling VoIP traffic (which is typically UDP) over SSL. There's a white paper in the Citrix Knowledge Center that explains how to enable the Cisco IP Communicator softphone through Access Gateway.

So, stay tuned for future blog posts where I will elaborate on each of these currently available methods for delivering telephony functions to your users via Presentation Server. And then I'll update you on where all of this might go in the future. Meanwhile, I'd be very interested in your comments on any of these solutions, the 'use cases' you believe are most important to address, and your own strategic thinking around delivering telephony functions to the enterprise.

Derek Thorslund
Product Strategist, Multimedia Virtualization

We recently conducted some tests to confirm that Office Communications Server 2007 can be delivered via Citrix Presentation Server 4.5. While these are not "official" test results, I thought many of you might appreciate an early look at what we found in case you're considering rolling out OCS 2007.

Office Communications Server, the successor to Microsoft Live Communications Server 2005, is Microsoft's entry in the Unified Communications space. It brings together Voice-over-IP (VoIP), Instant Messaging (IM), audio and video conferencing, and integration with Microsoft Office. OCS includes presence information so you can see at a glance whether someone is available to receive your phone call or instant message.

We didn't test video conferencing. That would require USB webcam support on Presentation Server. Our focus was on the Instant Messaging and Microsoft Office integration features of OCS 2007.

We published the Office Communicator client on Presentation Server and successfully used its Instant Messaging and presence functions. OCS integrates presence information from multiple sources including the Outlook calendar and Out-of-Office Assistant. From an e-mail message in Outlook, you can view the presence information for each addressee and then initiate real-time communications from within the message without switching applications.

Office Communicator can also be used to control a physical telephone set. For example, you can instruct Office Communicator to place a call in your behalf and, leveraging your telephone system, it will ring your phone (office, home, or mobile) and then call the other party and bridge the connections. You can't yet use Office Communicator on Presentation Server as a pure softphone with voice-over-ICA; one of the reasons is that softphones need to open the audio driver more than once (ringtone/busytone, voice) and the current audio driver in PS 4.5 FP1 doesn't support that. (We previewed an enhanced audio driver for softphone support and voice-over-ICA in the Tech Lab at iForum in October and I'll blog on various aspects of voice-over-IP in the new year.)

If you have any experiences running Office Communicator on Presentation Server that you'd like to share, please write a comment on this blog post. And I'll keep you informed as we learn more about delivering Unified Communications via Presentation Server and XenDesktop.

Derek Thorslund
Product Strategist, Multimedia Virtualization

Finding the optimal balance between audio quality and bandwidth consumption has never been easy. Most Citrix customers today choose our Medium Quality audio setting to obtain the best compromise for the majority of use cases. But in the Citrix R&D labs in Fort Lauderdale, our computer scientists are cooking up something much better. View my interview of Modesto Tabares, one of the lead engineers working on Citrix's Multimedia Virtualization Initiative, to learn about a new voice codec codenamed Ulysses that offers a win-win proposition.

Derek Thorslund
Product Strategist, Multimedia Virtualization