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The following are brief descriptions of the features of Integrated VoIP audio:
Services Support: Integrated VoIP is supported in the services and platforms listed in the following table:
Center |
Windows |
Solaris |
Linux |
Macintosh |
---|---|---|---|---|
Meeting Center |
Yes |
Yes |
Yes |
Yes |
Training Center |
Yes |
Yes |
Yes |
Yes |
Event Center |
Yes |
Yes |
Yes |
No |
Support Center |
Yes |
Yes |
Yes |
Yes |
Number of attendees: Integrated VoIP supports up to 500 attendees (1,000 for Training Center).
You can invite up to 500 attendees to a session (1,000 for Training Center).
Integrated VoIP displays a network indicator in the Volume window (available from the Audio menu) that shows how your network is performing and the overall quality of the audio your attendees hear. The indicator displays one of the following colors:
Your system must meet the requirements shown in Cross-platform Features (WBS30).
You can use Integrated VoIP with the WebEx services and computers listed in the following table:
Center |
Windows |
Solaris |
Linux |
Macintosh |
---|---|---|---|---|
Meeting Center |
Yes |
Yes |
Yes |
Yes |
Training Center |
Yes |
Yes |
Yes |
Yes |
Event Center |
Yes |
Yes |
Yes |
No |
Support Center |
Yes |
Yes |
Yes |
Yes |
To use WebEx Integrated VoIP, you will need a full duplex sound card and speakers or headset. To speak, you should have a microphone that is connected to your computer. For best results, we recommend that you use a headset.
You can use the UDP or TCP protocols with WebEx VoIP audio. With UDP, you may experience lower latency rates (delays) than with TCP, but with TCP, you can use the SSL security protocol (and the latency rate will probably be greater). When VoIP starts, WebEx tries to connect using UDP and then switches to TCP. You can conduct sessions where some attendees use UDP while others use TCP.
UDP is only supported for non-SSL sites. In order to use UDP, the IP ports 9000 and 9001 must be opened for outbound communication using UDP on the corporate firewall. UDP is selected automatically if the ports are open.
Yes. You can use SSL if you also use the TCP transport protocol.
Integrated VoIP is not recommended for dial-up connections. UCF-based PowerPoint sharing should work satisfactorily as long as video is not enabled and only one active microphone is in use. Application and desktop sharing in concert with Integrated VoIP is not supported on connections of less than 56Kb/s.
Integrated VoIP can be provisioned from a WebExTM Extended MediaTone eXchange (EMX) node on a case-by-case basis. Please contact Product Management for further information.
Integrated VoIP is full duplex, meaning multiple attendees can speak at the same time. This is similar to a traditional teleconference using PSTN. Half duplex is a VoIP conference where only one attendee can speak at a given time, similar to a CB radio.
Traditional PSTN-based teleconferencing is circuit-based, which gives each participant a dedicated channel to the teleconference bridge; the delay is virtually unnoticeable. Typically, the only delay one encounters in a circuit-switched voice environment is due to the distance the voice must travel). A good VoIP solution will have delay of about 0.25 - 0.5 seconds, depending the following factors:
Such delay and audio quality issues are common to the VoIP solutions offered by all the vendors—not just WebEx. VoIP solutions offered by vendors such as Centra, et al., suffer from the same shortcomings when compared to PSTN. Based on our testing, the delay and audio quality of WebEx VoIP is at least on par with that of Centra's.
It is hard to have a straight answer to this question due to the number of possibilities. You can have a perfect VoIP conference with a 28-Kbps connection to a country halfway around the world, followed by a scratchy mess for a call to the next state with a 56-Kbps or a 300+-Kbps connection. The quality is almost entirely determined by the sample rate (number of "slices" per second it uses to reproduce your voice) of the VoIP software, plus the throughput of your internet connection. A 56-Kbps (or a 300+-Kbps LAN, for that matter) connection does not ensure that you can move data across the Internet at that speed; the actual speed is determined by traffic levels on all networks between the source and end point, and the equipment capabilities at the source and end point. In general, poor-quality transmissions are a result of traffic and cannot be avoided completely in VoIP that uses Internet for all or part of the voice-data traffic.
Follow the standard Technical Support escalation process.