I am relatively non-technical so please bear with me as I explain my situation and the question that I would like some feedback on.
We have a Cisco video conferencing system consisting of a core infrastructure made up of a single instance of a TMS, VCS, IPVCR, and MCU. We currently have a mixture of older Tandberg Edge units, as well as newer SX-20, EX-60 and EX-90 units; the total number of units on the system is c180. The system sits inside a private, protected network to provide secure communications to select customers.
One of our customers has asked for voice-only connectivity over the network so that they can have a lower-cost secure comms option at the desk to sit alongside the vc units in their meeting rooms. Our first thought was to use SIP phones over the VCS knowing that it supports SIP, as we are probably only looking at 40-50 phones initially and we don't want to invest capital unless there is a demand for this service.
I apologise in advance as I have several questions that I am hoping you can answer:
Can we use voice-only SIP phones on the system with the current core infrastructure?
Would they be able to connect ad hoc point-to-point or are the phones registered in a different way to vc end-points?
What would be the limiting factor in terms of number of phones that we could connect together through the network?
Would the SIP phones be able to join in with multi-way video conferences with voice only?
Is there a better way of providing a voice-only network with the minimum of capital investment up-front and if so, what would that be?
I have asked our techies but they are used to dealing with video-conferencing setups and have no experience of connecting phones to the system. Any help that you could provide would be most welcome. If you need me to provide more detail to help you answer any of the questions, please let me know.
We run a bunch of Cisco 7960G IP phone that utilise a SIP image that register with various VCS Controls, so the bottom line answer to your question is, yes it can be done - in fact any standards SIP based device (including softphones and other video clients) should work to some degree. But there are caveats.
Our 7960G phones (for instance) are old, and the last SIP image for them was released in 2008. I can only get them to register with a URI of "name@ip_address_of_VCS" rather than "name@domain", however, we can match an alias and I get them to register in a separate subzone - just for clarity .
We utilise something called the GDS (Global Dialling Scheme), which allows H.323 devices to use a structured E.164 numbering scheme to dial other devices worldwide. Our SIP phone are set up to fake this E.164 number by registering something like "Fake_E.164@IP_Address_of_VCS", and we use search rules and interworking to strip and manipulate the search strings to enable calling between all units. Essentially, we utilise the SIP phone to provide simultaneous translation for a live videoconference - i.e. a completely separate call - although it is possible for a SIP phone to dial a VC unit and vice versa.
Of course there are drawbacks. Firstly, the phone uses UDP, which means we have to open up SIP for UDP on the VCS's. You will see a lot of post on here relating to SIP Spam scanning that mainly used UDP and the new VCS software has SIP UDP switched off by default.
Then there is the issue of call licensing (probably registration licensing will be OK). The VCS's are not really designed for voice SIP based products, and you might find that a lot of audio only endpoint consume the precious and expensive licences.
Still the underlying concept means that technically you can achieve what you want. I'm sure others will be able to provide further info.
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