Audio conferencing capability

hi all,
is it possible to bring audio conferencing kind of capabilities into portal

Hi this is henri
Please clarify as in do you have a third party application you would like to integrate which has audio conferencing capability??
Even better what is the scenario or requirement??

Similar Messages

Audio conferencing

I am in Skype when I go to WebEx to create a meeting and get participates added. There is an option for me to set up my Audio Conferencing using either my phone or my computer. Computer works fine. And my cell phone works fine. But when I try to use my Skype number, it always fails to connect. Is that because my Skype number is busy because I am on it already?

Hello,
Please, if anyone has any hint on this I would really appreciate. I have managed to do part of the audio conferencing, but, all I hear is garbage sound.
It�s not really garbage, it�s the sound received, but impossible to hear correctly.
The reason for this is that when I transmit RTP packets, I have a loos rate of 66%, which of course, causes the audio being sent to be unhearable. But I can remotely distinguish different sounds of different words, but of course, reconstruction of a speech with a loss rate of 66% is impossible.
I know that jmf has the limitation of getting a loss rate when transmitting of 16 - 17 %, but that is OK for understanding voice.
Any hints on how to solve the problem of the 66 % loss rate?
I read somewhere I cannot remember now that there�re tools for searching for busy waitings in programs and so on. Can you suggest me something to fix the problem?
Thanks and bye.
bgl

Problem in audio conferencing using Flex and FMS3.0

Hi Everyone,
I am trying to achieve audio conferencing using Flex and
FMS3.0
Currently I am following an approach where all the persons in
voice conference publish their voice on flash media server using
NetStream.Publish() function and all persons listen to the streams
published by all other members in the conference using
NetStream.Play() function. I am using a shared array on server side
to keep track of all the persons present in the conference. The
problem I am facing is that sometimes all the persons are able to
hear the voice of each other but other times audio of some persons
is not audible to others. Even if voice comes then there is echo
and one can hear the sound 3-4 times depending on the amplitude of
the sound. I have also used Microphone.setUseEchoSuppression(true)
to suppress the echo but it is not effectively suppressing it. If
someone has faced similar problems in past please let me know the
solution. Thanks in advance.
Regards,
Pankaj

I know.. but it was wrapped as jar or war files..
I searched all over my harddrive to find something named like org.apache or jdbc.clientDriver...and so on..with no result.
and I unziped derby.jar or derbyclient.jar etc.. also with no result.
It's like the driver has hiden somewhere like viruses...at least to me..
Edited by: dewyone on Dec 25, 2007 1:48 PM

Does iOS 8 FaceTime have audio conferencing or not?

In my searches on the internet for the audio conferencing feature in FaceTime, I find unofficial claims that it was to be included in iOS 8. And I see a button in the FaceTime interface to merge calls. But I have not found any explicit instructions on how to use this feature if it is in fact available. Is it? If so, what is the procedure to make us of it?

Same problem. I have tried to test FaceTime audio conferencing with two other patient iPhone 6 users (brother and sister) without success in the following steps.
1. We each turned off cell data and insured we had strong wifi.
2. I connected to one party and waited until the + icon appeared to dial in the second.
3. After connecting to the second party, the swap icon (horizontal S) was available so I could switch back and forth between parties, but the Merge icon (inverted Y) would not activate to bring all three of us together.
Has anyone had success with a three way conference call using FaceTime audio?

Large Scale Audio Conferencing Solution

Hi all,
I am researching into an audio conferencing solution that can support up to 150+ concurent participants. Anyone has the infrastructure in place to support this many concurent participants ? please share what you're currently using.
Thanks in advance !!
Danny

Hi Danny,
If you are interesting in on-premise Audio Conferencing solution, Cisco offers Cisco Unified MeetingPlace 8.5 solution. You can find out more about this solution here:
http://www.cisco.com/en/US/products/sw/ps5664/ps5669/index.html
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6789/ps5664/ps5669/data_sheet_c78-693132.html
I hope this is the information you are looking for.
Kind regards,
-Dejan

MPX audio conferencing

Hi all,
My customer has a 2 nodes CM cluster and a MPX (also CUPS and CUPCs...)
The point is that I'm trying to do audioconferences and the parties, located at differente sites, speak a mixture of G.711 and G.729.
For this reason, CM CFB doesn't work.
So I thought of using MPX to do the audioconferences.
To be honest, I don't know much about MPX, I just tested videoconferencing and some web collaboration, and we usually configure conferencing resources into routers to overcome this sw conference limitation, but as I saw that the MPX could do audioconferences I thought it was worth it to give it a try.
But it doesn't work. If all the parties speak G.711 it works fine, but as soon as one of the parties speak G.729 the conference drops.
As I said, my knowledge regarding MPX is very limited (and my ability to find what I was looking for in the documentation too :-), but I saw that you could specify the number of ports reserved for G.711 sessions and for G.729 sessions (and I did it so), so I thought ithe MPX could do transcoding if necessary.
Am I right? Should it work and I'm just misconfiguring the MPX?
Or do I need to configure transcoding in routers anyway?
Thanks in advance.
Best regards,
Carmen

Hello,
Please, if anyone has any hint on this I would really appreciate. I have managed to do part of the audio conferencing, but, all I hear is garbage sound.
It�s not really garbage, it�s the sound received, but impossible to hear correctly.
The reason for this is that when I transmit RTP packets, I have a loos rate of 66%, which of course, causes the audio being sent to be unhearable. But I can remotely distinguish different sounds of different words, but of course, reconstruction of a speech with a loss rate of 66% is impossible.
I know that jmf has the limitation of getting a loss rate when transmitting of 16 - 17 %, but that is OK for understanding voice.
Any hints on how to solve the problem of the 66 % loss rate?
I read somewhere I cannot remember now that there�re tools for searching for busy waitings in programs and so on. Can you suggest me something to fix the problem?
Thanks and bye.
bgl

FME 3.0 Audio from Audio Conferencing Bridge issue

All Hi
I am having an issue pulling audio from an audio conferencing bridge and then streaming it out via Flash Media Encoder 2.5. When I take another source, such as music or pre recorded voice, it sounds fine, however as soon as I switch to the audio from the audio conferencing bridge, the resulting output via the website that I stream to sounds very poor. Not awful but not good. Obviously I understand that the audio isnt going to be as good as you would get from an MP3 or pre-recorded file but I dont think it should be this bad.
Could it be something to do with my encoding settings.
I am using Mp3, Mono, Sample Rate of 44100 ( which is high I know ) and a bit rate of 96kbps.
I dont believe its my audio hybrid ( Sonifex HY-03 ) or my converter box ( stereo phono to firewire converter ) as all other audio going into FME is fine, its just this darned audio conferencing feed.
Could my encoding settings be too high?
Thanks in advance.

To configure Osprey 440 you need to select exact Input in FMLE from this device.
Osprey 440 has 1 audio slot and 4 composite Video slot.
Starting from Audio cable, you have Video A, video B, video C and video D.
Similarly on Audio breakout cable, you have right and left channels for each of the 4 inputs starting from left 1 right 1, left 2 right 2 and so on.
In FMLE these device are listed as Osprey 440 Video device 1A, Osprey 440 Video device 1B and so on.
You need to select exactly that input in both audio and video drop down list whom you are giving input in the card.
Hope this helps.
Let me know if you still face the problem

LR4: any AUDIO editing capability - simple (?) enhancement request

First, hats off to the Adobe team for a nice upgrade. I'm still getting used to the new slider layout, but am loving it already. Really nice to be able to pull back the highlights without the image going to crap.
In addition to shooting stills with my Canon 7D, I also shoot the occasional video. I'm just a hobbiest (so don't have the budget for Adobe Premier) but just like to do some simple edits. The new video capability in LR4 is very much appreciated ... but I can't seem to find any audio editing capability - I suspect none exists.
As a software developers, I know the LightRoom team is pretty busy with bug squashing on the just released LR4 ... but a very useful feature would be the ability to export with/without a sound track, merge with an external track, or discard one of the stero tracks and convert the other to a mono signal.
You might ask why would you want to do the later? The reason is the *&^%$#@! Canon 7D (and many other DSLR's) use AGC (Automatic Gain Control) for their audio ... so it basically sounds like crap as the levels wander up and down. Fortunately, there are some 3rd party amps (placed between the microphone and camera) that can "defeat" the AGC by inserting a signal in one of the stereo tracks ... and this way you get consistant levels. So right now, I have to open up the video file with Audicity (which pulls the audio) then delete the one track and use the other as a mono signal. I then save that WAV file, open the video back up in Quicktime Pro, delete the audio track associated with it, and then pull in/merge the previously saved WAV file ... and then finally exporting the combined video/audio.
It would be awesome to do all of my (simple) video editing in LR4, and then simply have an option to fiddle the audio output as shown:
     1. Leave Audio as-is.
     2. Combine Left and Right track into Mono Signal
     3. Delete all Audio
     4. Delete Left Track (optionally convert right track to Mono Signal)
     5. Delete Right Track (optionally convert left track to Mono Signal)
Again, the AGC "problem" is really because the DSLR manufacturers don't allow you to set audio levels on their cameras - this is fixed in the most recent models and also with Magic Lantern ... the later is unfortunately not available for the 7D. But my approach to "defeat" the AGC isn't uncommon and my hope (PRETTY PLEASE! ;-) is that this simple audio editing capability might be easy to add to Lightroom and worthy of consideration.
Thanks,
alek
P.S. Note that I don't see this impacting sales of Adobe Premier since professional videographers are using off-camera sound and are already buying that program. I.e. this is mostly an issue for us hobbiests that are trying to improve the sound quality on the cheap and Premier is total overkill for this - be awesome if LightRoom could help us out a bit.

Hi,
why must I be editing a package to use the find function!? Has been asked for on numerous occasions, and has been agreed to implement in a future drop.
why doesn't the "describe table" functionality work from within a packageThis too has been asked for, and agreed to be implemented in a future drop.
If you'd have read the sticky moderator posts, especially "Feature Requests", you'd have found this on the SQL Developer Exchange ;-)
K.

Video conferencing capability

Will the iphone have video conferencing capability or does it now?

It currently does not and we don't know if it will in the future.
You can send feedback - http://apple.com/feedback

IChat 4.0.2: audio conferencing not working

First of all, this question regard iChat 4, not "iChat AV 3" as the forum might suggest. It seems that Apple as of yet hasn't yet created a proper forum for iChat 4.
I'm using 4.0.2 on 2 machines. Machine A has been set up 2 weeks ago, machine B yesterday, no settings changed whatsoever. Problem: Audio conferencing doesn't work. Everything else (ping, webbrowsing, email, ichat text, ftp, sftp, ssh, skype) works perfectly fine. Both machines were connected directly to the cable modem.
What I've tried out so far:
- repeated login/logouts
- restarting the machine
- setting qt stream speed to 1,5 mbit and ichat speed to 500 kbit, as somebody suggested
- set firewall from "all" to "only certain programs" and added iChat, as somebody suggested
- changed server port to 443, as somebody suggested
- tried conferencing with the test accounts appleu3test01/02/03
- created a new user on the same machine, used it
- used the other machine which was basically out of the box
- inquired at the cable company whether they filter anything, they don't.
- submitted half a dozen error reports to Apple.
I'm close to start being a teeny weeny bit annoyed soon...
Has anybody got other stuff that I might try out?
Thanks

I have been having no video or audio conferencing problem using iChat 4.0.2 as have 5 of my friends. Further two of us tried to access the Apple tests sites with no success, the same connection error problem.
I called AppleCare today and the problem remained with every test situation. AppleCare sent a Support Document, which is aimed at OS X 10.4, and was not useful.
While looking around I may have found some helpful information. I had "Internet Sharing" on so that I am sharing Airport to Ethernet. When I turned that off Walla!! iChat with two different Apple Test Sites worked.
Of interest is that iChat worked between two MacBook Pro's on the same Airport Extreme and between a MacBook Pro and Intel iMac using a Ethernet to Airport Internet Sharing configuration.
I hope this is useful information for those who are frustrated.
Jim

Audio Conferencing using Cirrus?

Hi there -
I'm interested in building a small website for my company's development team. This is the scenario that I would like to achieve:
1. Colleague enters website
2. Colleague presses "Connect"
3. Colleague is connected with any other colleagues that have pressed "Connect"
4. Any colleagues Connected can chat using only a microphone I do not wish to support video feed
5. Full duplex communication is supported between all colleagues
Is this something I can achieve using Cirrus? If not, do you have any idea what technologies exist that could make this achievable?
I have found the following website:
http://www.speakfreely.org/
This seems to do exactly what I want, but I need to create a web service out of this without forcing my colleagues to download something to their computer.
Any tips would be appreciated. Thank you for your time!

My ReelPortal app does just that, except it also have video (which you don't have to turn on). Here's a short description:
ReelPortal is a video chat and conferencing software. Designed to work with multiple devices and OSes (Windows, Linux, MacOS, Android, BlackBerry Tablet, etc), its features include:
* Video conference with multiple friends
* Create your own private rooms
* Supports both p2p and client/server connections
ReelPortal is room-centric. You automatically connect to everyone in a room when you enter it. No need to call them individually.
You can run it directly from the web browser on the PC/Mac/Linux, at www.reelportal.com. You can even download the server, and run it within your own intranet. But if you don't like it, you can certainly implement your own via Flash/AIR.
As for audio quality, using headset is probably the best, but if you really need speakerphone, I'd suggest get a decent Bluetooth Speakerphone, like the Motorola EQ7 or EQ5, at a very affordable price.

Audio recording capability

So, why can't I find more about the vpod's audio recording ability? Seems this capability will make it a must have tool for podcasting, news reporting, musicians, etc. Apple seems to barely mention it. Is it for real? How can I plug in my mic? How about battery life while recording? Is it time to give up my minidisc recorder?
john at woodentoy.com

From the manual, hope it helps:
Recording Voice Memos
You can record voice memos using an optional iPod-compatible microphone (available for purchase at www.apple.com/ipodstore). You can store voice memos on your iPod and load them on your computer. You can set iPod to record at Mono quality (22 kHz, mono) to save space, or Stereo quality (44.1 kHz, stereo) for a better sound.
To record a voice memo:
1 Connect a microphone to the Dock connector port on your iPod.
2 Set Quality to Mono or Stereo.
3 To begin recording, choose Record.
4 Hold the microphone a few inches from your mouth and speak. To pause recording, select Pause.
5 When you finish, select Stop and Save. Your saved recording is listed by date and time.

VOIP + Integrated Audio Conferencing + Connect 8

Hello -
These forums have been a valuable resource for my individual troubleshooting in Adobe products, and I thank the contributors.
Currently I am having an issue that I am unable to resolve on my own, nor find any postings covering simlar topics through search.
Core Question:   Can one create a room in Adobe Connect 8 that does all of the following?:
Allows virtual attendees to connect through VOIP
Allows virtual attendees to connect by selecting to have the integrated conference line call them (where they must enter their phone number), where that attendee can be muted individually, their audio status is displayed by their username etc.
Allows (and provides directin for) the attendees to dial into the conference line
Specs:
Currently Adobe Connect 8 (Service Pack 1)
PGi Conferencing Service
Adobe product support through Clarix
Background:
For our implementation(s), we need to broadcast many different event types and offerings to a virtual audience in which presenters/instructors and participants can share two way verbal communication with one another. As the host, I must command control of the audio at an individual level using an integrated conferencing service (we use PGi). VOiP users are already controlled at the individual level. We must provide the easiest and most flexible access to the Adobe Connect room for our virtual attendees, so that they can click a link and sign on as a guest and then choose whether or not they wish to use VOIP, dial into the conference line from their phone, or ask the Adobe Connect room to dial out to their phone.
We have successfully delivered conferences in the format that I describe using an Adobe Connect-based product, which I presume to be built on a previous version of Adobe Connect. This version had the little yellow 'dial out' phone button. Let me add that I am aware of and understand what Universal Voice is and how it works, and that Universal Voice will not meet our needs. The latest and greatest Adobe Connect 8 Help PDF had this to say on page 126 about using VOiP and Conference together:
http://help.adobe.com/en_US/connect/8.0/using/connect_8_help.pdf
     Note: If Universal Voice is configured, you can select both options. Participants can join the audio conference either using their computer's
               microphones, or join the meeting using their phone. Start broadcasting to to enable participants to hear any telephone-based audio through the
               computer speakers and broadcast their voices to telephone users using the microphone. When you stop broadcasting, VoIP users will be audible
               to phone users, but phone users will not be audible to the meeting users.
Anybody with insight please reply, and I thank you in advance.
Warm Regards,
:Otto:

Otto,
I'm baffled by this. Connect 8 does all what you desire with Universal Voice and PGi. I do all what you desire today. Connect 8 SP1 which is being rolled out changed the menu choices to make the options really clear. They were not clear in Connect 8 and many people got confused including me!
What is it about UV and the integrated PGI adaptor that does not work for you? Best to take this off-line I think as we will need to discuss.
You can reach me here: [email protected]
Thanks.

Audio conferencing through MSN?

Is there anyway to do this?

Install Parallels, Boot Camp, or Wintel OS/X on your Mac and run Windows. Some of these will support audio chat clients on Windows. See my WINTEL solutions section of my FAQ for the links to these various solutions:
http://www.macmaps.com/macosxnative.html#WINTEL

Audio conferencing Help

Hey guys,
My sister has an iBook. As you might know already, iBooks only have 1 audio line-in (dunno how you call them), which is the headphone input.
This is her problem. She is doing this language course with someone who is using MSN on a PC. She has MSN and Mercury installed on her laptop.
What she wants is to be able to AUDIO conference with her teacher, preferable with some sort of headphone so people around her won't be disturbed by his voice. She doesn't need to see him on video, but she has to be able to talk to him and vice versa.
iBooks have internal microphones right? She also has a bluetooth headset, but obviously that won't work.
I tried to help her and tried this:
I logged in on our PC and went on MSN. My sis got on her iBook and went on Mercury. She has an iSight and so we did a videoconference. The problem was that we couldn't talk to each other straight away. I could see her on her webcam, but I couldn't hear her speak nor could she hear me talking through a microphone hooked on my pc. Why did her internal microphone not work, nor the microphone built-in with her iSight?
This is just one way I could think of to be able to do audio conference with someone on a PC using MSN.
What's wrong with my approach?
Does anyone else have other suggestions?
Thanks in advance!

MSN as far as I know doesn't support audio on the Mac. iChat will do audio chat with AIM on a Windows machine. Suggest they go about that method with the built-in microphone of the iBook.
For more on alternatives, see this page:
http://www.applechat.info/

Audio conferencing capability

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