Re: [asterisk-users] RTP Mixer
Anybody I am still waiting. Kapil Dhawan wrote: Hi Can somebody brief me the working of RTP mixer from MeetMe perspective. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil Dhawan wrote: Hi Can somebody brief me the working of RTP mixer from MeetMe perspective. (RTP? in MeetMe? I don't really know this well but AFAIK RTP doesn't get mixed directly in MeetMe. MeetMe is a generic Asterisk channels mixer) Aparantly people either don't know enough or don't have the time. Try rephrasing your question so it will be more specific and thus also hopefully take shorter time to answer. Do you have a working system? Do you need to set up one? What version of Asterisk? What types of channels do you try to mix? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
I was reading an article on RTP Mixer so started studying about the mixing done by Asterisk in MeetMe. Read that CC should contain the no of participants ifupto 15 and CSRC should come, but not getting any by asterisk. Tzafrir Cohen wrote: On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil Dhawan wrote: Hi Can somebody brief me the working of RTP mixer from MeetMe perspective. (RTP? in MeetMe? I don't really know this well but AFAIK RTP doesn't get mixed directly in MeetMe. MeetMe is a generic Asterisk channels mixer) Aparantly people either don't know enough or don't have the time. Try rephrasing your question so it will be more specific and thus also hopefully take shorter time to answer. Do you have a working system? Do you need to set up one? What version of Asterisk? What types of channels do you try to mix? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
Kapil Dhawan wrote: I was reading an article on RTP Mixer so started studying about the mixing done by Asterisk in MeetMe. Read that CC should contain the no of participants ifupto 15 and CSRC should come, but not getting any by asterisk. I'll just leave it at this so we can all move on with our lives: Asterisk isn't totally an RTP Mixer in the sense you are reading about. It is an audio mixer. Frame of audio comes in over RTP, gets sent in (only the audio portion) to be mixed, frame comes out, gets turned into RTP again. The RTP part has no idea that multiple sources were mixed together, 'nor should it care. The sources could have been Zaptel channels for example in which case they couldn't be added to the list. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
Perfect Josh...but if i am running an application which has a capability of showing number or participants depending upon CC value, that doesn't work. Secondly, Asterisk can show on CLI about current talking users where it is maintaining talking status but not sending it down the line to be used by other apps. Anyways, i will go with your statement and leave it on core developers to comment. Joshua Colp wrote: Kapil Dhawan wrote: I was reading an article on RTP Mixer so started studying about the mixing done by Asterisk in MeetMe. Read that CC should contain the no of participants ifupto 15 and CSRC should come, but not getting any by asterisk. I'll just leave it at this so we can all move on with our lives: Asterisk isn't totally an RTP Mixer in the sense you are reading about. It is an audio mixer. Frame of audio comes in over RTP, gets sent in (only the audio portion) to be mixed, frame comes out, gets turned into RTP again. The RTP part has no idea that multiple sources were mixed together, 'nor should it care. The sources could have been Zaptel channels for example in which case they couldn't be added to the list. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
Check http://bugs.digium.com/view.php?id=9384 On Mon, 7 May 2007 20:19:24 +0630, Akashdeep Dutta wrote Hi Does asterisk uses CSRC. I was trying to create a situation with 4 simultaneous users talking and executed ethereal to capture logs, studies RTP but never found CSRC count to be more than 0. Can somebody explain me. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
Does Asterisk really follow 3550 for CSRC. I am confused. On Mon, 7 May 2007 20:19:24 +0630, Akashdeep Dutta wrote Hi Does asterisk uses CSRC. I was trying to create a situation with 4 simultaneous users talking and executed ethereal to capture logs, studies RTP but never found CSRC count to be more than 0. Can somebody explain me. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
Akashdeep Dutta wrote: Does Asterisk really follow 3550 for CSRC. I am confused. On Mon, 7 May 2007 20:19:24 +0630, Akashdeep Dutta wrote Hi Does asterisk uses CSRC. I was trying to create a situation with 4 simultaneous users talking and executed ethereal to capture logs, studies RTP but never found CSRC count to be more than 0. Can somebody explain me. Asterisk isn't an RTP mixer in the sense you are thinking of. Asterisk mixes/exchanges audio frames in a protocol independent fashion. The mixed frame then gets sent into the RTP core where it gets turned into an RTP packet that is sent out. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
How does it really mix the packets. If i have three participants talking together and packets are reaching Asterisk, How does it mix them. Asterisk isn't an RTP mixer in the sense you are thinking of. Asterisk mixes/exchanges audio frames in a protocol independent fashion. The mixed frame then gets sent into the RTP core where it gets turned into an RTP packet that is sent out. Joshua Colp Software Developer Digium, Inc. On Mon, 7 May 2007 21:10:12 +0630, Akashdeep Dutta wrote Does Asterisk really follow 3550 for CSRC. I am confused. On Mon, 7 May 2007 20:19:24 +0630, Akashdeep Dutta wrote Hi Does asterisk uses CSRC. I was trying to create a situation with 4 simultaneous users talking and executed ethereal to capture logs, studies RTP but never found CSRC count to be more than 0. Can somebody explain me. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
On Tue, 8 May 2007, Akashdeep Dutta said something to this effect: How does it really mix the packets. If i have three participants talking together and packets are reaching Asterisk, How does it mix them. I imagine that to a large extent the payloads are simply superimposed, possibly with some sort of duplex considerations akin to traditional conference bridges, in much the same way that most DSP drivers for audio boards capable of muxing multiple input streams do. There is no doubt a codec-based / waveform explanation that is difficult to pinpoint without considerable expertise in that area of engineering. -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
As I understand it, if you are using MeetMe, or if there is any codec translation, or having to listen to DTMF (it should only have to decode the audio to listen to INBAND DTMF, I don't know if it does), or other sorts of things that requires Asterisk to listen to or manage audio the audio will be decoded into SLN format. I believe for MeetMe all the audio channels are decoded into the SLN format then that audio in SLN format is handed to the Zaptel driver to use some sort of voodoo magic to mix the audio. Asterisk does not touch the actual audio except for after it has been converted to SLN format. Akashdeep Dutta wrote: How does it really mix the packets. If i have three participants talking together and packets are reaching Asterisk, How does it mix them. Asterisk isn't an RTP mixer in the sense you are thinking of. Asterisk mixes/exchanges audio frames in a protocol independent fashion. The mixed frame then gets sent into the RTP core where it gets turned into an RTP packet that is sent out. Joshua Colp Software Developer Digium, Inc. On Mon, 7 May 2007 21:10:12 +0630, Akashdeep Dutta wrote Does Asterisk really follow 3550 for CSRC. I am confused. On Mon, 7 May 2007 20:19:24 +0630, Akashdeep Dutta wrote Hi Does asterisk uses CSRC. I was trying to create a situation with 4 simultaneous users talking and executed ethereal to capture logs, studies RTP but never found CSRC count to be more than 0. Can somebody explain me. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
