LanScape Support Forum: controlling mic input destination in cases of multiple lines

		Active Topics Member List Search Help Register Login

LanScape VOIP Media Engine™ - Pre-Sales Technical Support
LanScape Support Forum -> LanScape VOIP Media Engine™ - Pre-Sales Technical Support

Topic: controlling mic input destination in cases of multiple lines

Author

Message

<< Prev Topic | Next Topic >>

ramon
Intermediate

Joined: April 04 2007
Posts: 1

Posted: April 05 2007 at 7:56am \| IP Logged

Hi,

We're interested in redesigning a H.323 based application to use VoIP, and are currently evaluating your product. Two questions:

1) One of the application requirements is to listen to multiple lines but send the mic input only to one. Is there (or will there be) a way to do this with your engine? This level of control seems somewhat related to the multiple-conferences support discussed in:

Multiple Conferences on multiple lines:
http://www.lanscapecorp.com/forum/forum_posts.asp?TID=144&PN =4

2) Do you guys have any thoughts about H.323-to-VoIP bridges? There seems to be many software and hardware-based solutions available, so I was wondering what kind of experiences you might of had with them.

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: April 17 2007 at 1:52pm \| IP Logged

Hi ramon,

Thanks for your post and for waiting for us to catch up from last week.

Item1: Listen to multiple lines and send mic data out only one of the lines:
I think we understand your question. A real simple way to accomplish this would be to have your VOIP app request that RTP media packets be sent to your app from the media engine. This is accomplished by having your app call the EnableRawRtpPacketAccess() API proc. The media engine will then send all ready-to-be-transmitted and received RTP packets to your app before then media engine processes them.

You can allow all received RTP media packets to be handled as normal. As far as RTP media packets that your app wants to transmit, your app can determine on a per phone line basis if RTP media should be transmitted out the individual phone lines. This would be done in your RTP_CALLBACK_PROC handler. In your callback handler, you would test for transmitting RTP payload packets by looking at the contents of the RAW_RTP_DATA structure that gets passed to your app’s callback. If you do not want the RTP media to be transmitted out a specific phone line, your app can set the ProcessRtpPacket structure member to FALSE (zero) and the media engine will not transmit the RTP packet for the line.

In the above scenario, the media engine RTP packet access allows you to perform a basic mute of all transmitted RTP payload on a per phone line basis.

Item2: H.323-to-VoIP bridges:
Sorry ramon. We do not have any info for you regarding h323 - SIP/RTP bridges. We strictly deal with SIP/RTP here.

Repost as needed,

Support

If you wish to post a reply to this topic you must first login
If you are not already registered you must first register

Printable version

You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot delete your posts in this forum
You cannot edit your posts in this forum
You cannot create polls in this forum
You cannot vote in polls in this forum