[cisco-voip] Streaming multicast audio source to an IP phone

[Stephen Welsh]

Hi Tim,

>From what you describe all you need is a static XML Page that would allow a user to join/leave a multicast session, so you are in the right place of the SDK, if you have a static page with something like the following hosted on a website:

<CiscoIPPhoneMenu>
  <Title>Broadcast Service</Title>
  <Prompt>Please select an option</Prompt>
  <MenuItem>
   <Name>Join Session 1</Name>
   <URL>RTPMRx:239.0.0.10:20480:100</URL>
  </MenuItem>
  <MenuItem>
   <Name>Leave Session 1</Name>
   <URL>RTPRx:Stop</URL>
  </MenuItem>
</CiscoIPPhoneMenu>

Note: If you host something like this on a website, make sure the Content Type header is set to "text/xml" or it won't be parsed by the phone correctly. You could make this more dynamic etc, but it's a simple way to do it.

There are additional methods to start receiving the audio session if the codec is something other than G.711ulaw, have a look here:
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/all_models/xsi/8_5_1/componentapis.html#wp1036347

Also,
I recommend you download PhoneView from Unified FX (http://www.unifiedfx.com), it does a lot of things, but one feature that would help your testing is the ability to send ANY Internal URI to an IP Phone, so for example you can select one or more phones, enter the "RTPMRx:239.0.0.10:20480:100" command and click "Send".

Thanks

Stephen

On 24 Apr 2012, at 21:17, Tim Reimers wrote:

Hi everyone-

I’m researching how to possibly do something with IP phones and multicast—

We want to stream audio from meetings in a very plug-and-play user-friendly way,
to IP phones and also PCs on the network.

I will be locating some type of hardware that can be connected to the line out of an audio system to pick up the audio, and provide a
multicast stream.
The multicast stream will NOT be coming from an IP phone –
It will be something like a PC with VLC loaded on it, or with proof of concept to get some funding, I might use a VBrick or something like that.

We want it to be a black box that a non-technical user can just switch on along with the audio system for the meeting room, and streaming
will automatically begin.

On the receive side, we want users to be able to listen to the stream on an IP phone or a PC, without having to
do much more than select a service or dial a number on the system, or click a link with a PC.

This is a one-way listen to the meeting; there is no need or requirement for audio from the client side, so the IP Phone should automatically go into speakerphone mode with mute, or allow audio to handset/headset

I have seen the below from the IP Phone SDK – I don’t know if I could leverage that into a service that is
pre-programmed to join a multicast “conference” (aka audio stream, for my purposes)
on a known multicast IP.

Thanks for any advice you guys have!

I know sometimes I forget to come back in here and thank you for times you give me great assistance-
Along those lines, major thanks to Peter Snow and others who assisted me with
echo cancelling questions and PRI setup!

14. MConference (JSP)
<image001.gif>

MConference is a sample application which allows users to create, join, and control IP multicast conferences
from an XML phone service. It's a good example of the power of XML services and IP multicast - providing a large-scale (thousands of participants), arbitrated conference using a simple web application.
Conference arbitration can be enabled/disabled by the conference owner. When arbitration is enabled participants must request permission which is then granted by the conference owner. When conference arbitration is disabled, participants can simply push-to-talk, similar to a hoot-and-holler network.






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