Posted: Tue May 26, 2009 8:51 pm Post subject: [Asterisk-video] Can the current Asterisk w/ video meet our
Jamie Clark a écrit :
On Tue, 26 May 2009, Emmanuel BUU wrote:
Disclamer: our company is operating a video call center for a French VRS
/ VRI operator (Websourd).
We have a dedicated product line for VRS /VRI based on Asterisk.
Nice! It might be something that we are looking for...
Jamie Clark a écrit :
We are looking for ACD/PBX solution that will meet our needs.
We have two groups of callers:
1) ==> A group of deaf callers using H.323 videophone (95% or so of the
installed base) and SIP videophone. H.263 is the default video codec.
Asterisk chan H.323 does not support video calls. It could be added or
you need to have some H323 to SIP gateway.
What is recommended H323 to SIP gateway? Is that only for call signaling only? What about RTP? Is media (i.e. voice/video codecs) done at endpoints only? (i.e. not having to go through the gateway which will surely overload the server with some number of ongoing sessions.)
Depends if you want to handle NAT traversal for H.323 (which is VERY tricky). Basically, we would setup a gatekeeper with some RTP proxy capabilities and
have either an H.323 capable asterisk configured as an H323 neighbour or only a H.323 to SIP gateway without RTP relay.
In all cases, RTP media would have to flow trough one server to handle NAT traversal in a reliable manner. Furthermore, if you need full blow Video Relay Service, you will need to mix one video conversation with an additional audio leg. To that, the media need to be brought to an Asterisk server.
2) ==> A group of hearing callers using voice only phones (cell phones,
PSTN phones, VOIP softphones).
We have a call center where relay agents take incoming video call from
deaf callers and then dial out to hearing callers using voice call and
The relay agent does the translation from ASL signing into spoken word and
Yes with some tricky dialaplan and a modified version of app_confernce.
Nice, as long it works :) We also like to set up conf call time to time.
(Both voice and video).
Then you need a full blown MCU.
The call center will need to have ACD setup so either incoming video/voice
calls will be routed to available relay agent. If there is no available
relay agent then display the video saying "Waiting for next available
relay agent" (for deaf callers) and recorded message saying "Waiting
for..." for hearing callers.
The deaf callers will use their H.323 based videophone to make calls to
call center using IP address i.e. md.callcenter.com.
We will have several call centers set up around the country, i.e. in MD,
PA, VA, IN, NY etc... If deaf video caller dial callcenter.com (only an
example), the ACD will route the call to first available relay agent
anywhere in USA. If deaf video caller dial va.callcenter.com then the ACD
will attempt to route to first available relay agents in VA call center,
if none exist then route to next group of agents in MD and so on.
Yes. One can define a cascade of queue
Learning some new words as I am new to telephony system :)
Other key needs of the system are:
1) Detailed call logging stats/reporting for both video/voice calls,
reporting for specific relay agent, call center supervisors and national
Yes using an external reporting app.
Nice as I am sure all of the data is being stored in database such
THis would be the case.
2) Relay agent should be able to transfer both voice and video call to
other available relay agent (for many reasons such as needing a break,
other agent can do better service for that call, etc...) anytime.
I would guess the relay agent's station will have two softphones, one for
voice only and other one for video running on a single PC. I would assume
each PC will need to have two IP address: One IP for voice softphone and
other IP address for video softphone or use one IP address but use
different call signaling ports (i.e. 1720 and 1721 for H.323).
No, We have a solution where the agent has a sigle workstation and our
server do the audio / video mixing on the three legs.
Hmm, interesting... as long it works, fine with us :)
Can the current Asterisk system meet all of the above needs?
Definitly, but not the "official" version. We have a modified one. The
only issue with your project is handling H.323. But again, H.323 video
support can be added.
So the SIP video is fully supported right now and only thing needs to
be added is H.323 video which seems not a big deal, correct?
Do I need anything else to make it happen?
Is there any good company that can help us set up this system?