Примечания к выпуску для Cisco Cisco WebEx Event Center WBS29.8
Integrated VoIP Audio FAQs
38
WebEx Event Center Release Notes and FAQs
Training Center
Yes
Yes
Yes
Yes
Event Center
Yes
Yes
Yes
No
Support Center
Yes
No
No
No
Sales Center
No
No
No
No
Q. What do I need to use WebEx Integrated VoIP?
A. 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.
Q. Can I use TCP, UDP, or PSTN with WebEx Integrated VoIP audio?
Q. Can I use TCP, UDP, or PSTN with WebEx Integrated VoIP audio?
A. 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.
Q. Can I use WebEx Integrated VoIP if my site is SSL-enabled?
Q. Can I use WebEx Integrated VoIP if my site is SSL-enabled?
A. Yes. You can use SSL if you also use the TCP transport protocol.
Q. Can I use VoIP over dial-up connections?
Q. Can I use VoIP over dial-up connections?
A. 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.
Q. Can I provision WebEx VoIP from an EMX node.
Q. Can I provision WebEx VoIP from an EMX node.
A. Integrated VoIP can be provisioned from a WebEx
TM
Extended MediaTone eXchange (EMX) node on a
case-by-case basis. Please contact Product Management for further information.
Q. Is VoIP a full or half duplex transmission?
Q. Is VoIP a full or half duplex transmission?
A. 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.
Troubleshooting
Q. Why is there a delay in the audio during my VoIP conferences? Why does the quality seem to
be not as good as traditional telephony?
A. 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