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Communication, No encoding IP to IP
zerowalker
post Jul 13 2013, 15:19
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Okay the Title doesnīt really make much sense.

But what i want to ask is, is there a way to communicate like, for example, Mumble, but without any encoding (Opus, AAC or whatever).
Meaning, pure PCM going from one end to the other?

And doing this between 2 PCs, meaning IP to IP, No Server along the way (One of the PCs can act as a server of course), but no IP1 -> Server -> IP2, if you get it.


I know this is probably not what anyone really wants, but i would like to know if it exists, or is a easy way to do it?
I am guessing, as i donīt want any complicated things like, Encoding, Noise Reduction etc etc, it should be fairly easy.


But well, then again, i havenīt found anything.

Thanks

PS: A moderator, if you can come up with a better Topic, just change it, cause it doesnīt even make sense;S

This post has been edited by zerowalker: Jul 13 2013, 15:30
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julf
post Jul 13 2013, 16:30
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QUOTE (zerowalker @ Jul 13 2013, 16:19) *
Meaning, pure PCM going from one end to the other?


Not sure of why you would want to do it, but you could easily write a program that sends raw PCM from one PC to another. You probably need to implement your own synchronisation and error correction mechanism.

QUOTE
And doing this between 2 PCs, meaning IP to IP, No Server along the way (One of the PCs can act as a server of course), but no IP1 -> Server -> IP2, if you get it.


No, unfortunately at least I don't get it. What do you mean with "IP"? Clearly not Internet Protocol?

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zerowalker
post Jul 13 2013, 16:40
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QUOTE (julf @ Jul 13 2013, 17:30) *
QUOTE (zerowalker @ Jul 13 2013, 16:19) *
Meaning, pure PCM going from one end to the other?


Not sure of why you would want to do it, but you could easily write a program that sends raw PCM from one PC to another. You probably need to implement your own synchronisation and error correction mechanism.

QUOTE
And doing this between 2 PCs, meaning IP to IP, No Server along the way (One of the PCs can act as a server of course), but no IP1 -> Server -> IP2, if you get it.


No, unfortunately at least I don't get it. What do you mean with "IP"? Clearly not Internet Protocol?


Well sadly, my programming skills are close to Null:


And when i mean IP i do mean The IP Address. Meaning, that the only connection done, is from 1 Address to The Other, Excluding of course the routing going on from your Internet Provider.

But not like, Skype for example, there i will always have to use, Microsofts(i guess) server, which will add to the latency.

And why i want it, is cause, Bandwidth isnīt a problem, and latency isnīt either.

The one i will be connecting to is 3ms away in latency.
So if i encode the audio, the frame-size latency will be much bigger then the latency itself.

And yes, i know that itīs probably not noticeble and all that, but still i would like it.


But if i have to program is form scratch, then i have to give up sadly, i have no idea how to do that.
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soulsearchingsun
post Jul 13 2013, 17:27
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What's the purpose of this transmission? If you give details on how you would be using said connection, we could probably point you to alternatives. Is this local or internet (probably local with this ping)?
edit: typo.

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DonP
post Jul 13 2013, 17:31
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QUOTE (zerowalker @ Jul 13 2013, 11:40) *
But not like, Skype for example, there i will always have to use, Microsofts(i guess) server, which will add to the latency.


I haven't followed the details of Skype that much since the Microsoft takeover, but used to be that skype servers were only involved in hooking you up with your destination. Once that was done all the actual talk/video was straight between you and him. Might be different for conference calls.

Now I read that Msoft is helping Homeland Security monitor audio/video when ordered to, so apparently they've poked some holes in the encryption for compliance purposes. I guess you need a pre-microsoft version if you want privacy from that.
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zerowalker
post Jul 13 2013, 17:43
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Didnīt know that about Skype. But well Skype isnīt the greatest when it comes to Audio, though i love the chat.

And well, the purpose of this is simply, to talk to each other, thatīs it.

And itīs not Local.

But we both got Fiber, and live close to each other, so the latency is close to none, and i really want to use that to itīs fullest.
As bandwidth isnīt a problem between us, there is no need for compressed audio.
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saratoga
post Jul 13 2013, 19:37
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Theres literally hundreds of direct video conferencing apps.
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zerowalker
post Jul 13 2013, 20:11
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QUOTE (saratoga @ Jul 13 2013, 20:37) *
Theres literally hundreds of direct video conferencing apps.


I am not interested in Video, though it would be nice to have that, and a lossless codec.
But that is surely a lot more complex as i canīt see any software using that.


What i am looking for is just Audio chat, without encoding.
Just pure PCM going through.
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julf
post Jul 13 2013, 20:47
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QUOTE (zerowalker @ Jul 13 2013, 21:11) *
What i am looking for is just Audio chat, without encoding.
Just pure PCM going through.


And why don't you want any encoding?
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zerowalker
post Jul 13 2013, 20:51
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QUOTE (julf @ Jul 13 2013, 21:47) *
QUOTE (zerowalker @ Jul 13 2013, 21:11) *
What i am looking for is just Audio chat, without encoding.
Just pure PCM going through.


And why don't you want any encoding?



Cause it adds latency, which i donīt want in this case.
Else i would gladly use Mumble, even though they havenīt allowed settings it to 2.5ms.
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julf
post Jul 13 2013, 20:57
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QUOTE (zerowalker @ Jul 13 2013, 21:51) *
Cause it adds latency, which i donīt want in this case.


Using the network will add more latency than the encoding.
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zerowalker
post Jul 13 2013, 21:00
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QUOTE (julf @ Jul 13 2013, 21:57) *
QUOTE (zerowalker @ Jul 13 2013, 21:51) *
Cause it adds latency, which i donīt want in this case.


Using the network will add more latency than the encoding.


Not in this case i think.

The latency is 3ms between us.

Or well, sure Opus 2.5 is less than 3ms, but itīs like 75% the latency added.

And well, i just want it, but i clearly understand that itīs not very useful overall.
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DonP
post Jul 13 2013, 21:03
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You realize when you are quibbling about single digits of milliseconds it is like worrying about the latency if the other person was in the same room as you speaking directly from 5 feet away, right?
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zerowalker
post Jul 13 2013, 21:04
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QUOTE (DonP @ Jul 13 2013, 22:03) *
You realize when you are quibbling about single digits of milliseconds it is like worrying about the latency if the other person was in the same room as you speaking directly from 5 feet away, right?


Yes
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saratoga
post Jul 13 2013, 21:29
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QUOTE (zerowalker @ Jul 13 2013, 15:51) *
Cause it adds latency, which i donīt want in this case.
Else i would gladly use Mumble, even though they havenīt allowed settings it to 2.5ms.


The latency in the rest of your system will be larger than that anyway, so this reasoning is pointless.

QUOTE (zerowalker @ Jul 13 2013, 15:51) *
I am not interested in Video, though it would be nice to have that, and a lossless codec.
But that is surely a lot more complex as i canīt see any software using that.


Then don't use it?

Seriously, a couple minutes on Google and you could have something working ...
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[JAZ]
post Jul 13 2013, 22:11
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You don't find any solution to your problem, because your problem is not considered a problem that needs a solution.

You also forget other parts which add latency, namely the soundcard input/output.

Everything less than 20milliseconds is considered realtime. Your mobile phone is probably in the order of hundreds of milliseconds.

You could also consider a VoIP phone.

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zerowalker
post Jul 14 2013, 09:19
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QUOTE ([JAZ] @ Jul 13 2013, 23:11) *

You don't find any solution to your problem, because your problem is not considered a problem that needs a solution.

You also forget other parts which add latency, namely the soundcard input/output.

Everything less than 20milliseconds is considered realtime. Your mobile phone is probably in the order of hundreds of milliseconds.

You could also consider a VoIP phone.


Probably, damn, well it was worth asking for it.

And yeah of course, the soundcard, not sure how much latency that adds with WASAPI and the like, but is surely isnīt Zero thatīs for sure.

Not into knowledgeable about VoIP, but isnīt that just a normal phone that works with your IP instead?
Meaning, you donīt use your PC, just the Phone line (which is the same as the Internet line).

Though may be wrong about that, i donīt even have a phone except mobile.

Thanks for answering, much appreciation.
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julf
post Jul 14 2013, 13:10
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QUOTE (zerowalker @ Jul 14 2013, 10:19) *
Not into knowledgeable about VoIP, but isnīt that just a normal phone that works with your IP instead?
Meaning, you donīt use your PC, just the Phone line (which is the same as the Internet line).


Just trying to understand the terminology here... "IP" is either "Intellectual Property" (unlikely) or "Internet Protocol". Thus the phrase "Your IP" doesn't really make sense. "Your IP network", "your IP connection", "your IP address" are all possible...

Your internet connection might or might not use your physical phone line (using ADSL), but they are not the same thing. The actual copper line can be shared between your internet (ADSL) service and your POTS dialtone service, but that is all they have in common.

VoIP is transmitting voice over an IP connection, either from a computer or a dedicated phone device.
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zerowalker
post Jul 14 2013, 13:35
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QUOTE (julf @ Jul 14 2013, 14:10) *
QUOTE (zerowalker @ Jul 14 2013, 10:19) *
Not into knowledgeable about VoIP, but isnīt that just a normal phone that works with your IP instead?
Meaning, you donīt use your PC, just the Phone line (which is the same as the Internet line).


Just trying to understand the terminology here... "IP" is either "Intellectual Property" (unlikely) or "Internet Protocol". Thus the phrase "Your IP" doesn't really make sense. "Your IP network", "your IP connection", "your IP address" are all possible...

Your internet connection might or might not use your physical phone line (using ADSL), but they are not the same thing. The actual copper line can be shared between your internet (ADSL) service and your POTS dialtone service, but that is all they have in common.

VoIP is transmitting voice over an IP connection, either from a computer or a dedicated phone device.


Well my english is lacking, and what i am trying to say is simply.

PC to PC, a straight connection, send and recieve.

Like most chat or communication probably work.

And i donīt have ADSL, i got Fiber, and if i want to use a phone, i am pretty sure it will use my IP as the "phone number", then just translating it or something.
As you canīt use the actual IP as a phone number when calling.

Or well, you probably Can, but it would be hard to remember them.

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[JAZ]
post Jul 14 2013, 15:42
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Looks like I added more confusion than needed when talking about VoIP..

There are several technologies, and they just share the concept, not how they are provided (also, see footnote about ADSL).

Telephone over IP: An internet provider, if implemented, is able to provide a phone number (real) which is routed using the Internet Service. In this case, the phone is connected directly to the router (Any digital telephone. Not a VoIP one), and the signal sent over the IP protocol, which does NOT mean sending it over Internet (IP is just a way to send messages between machines of different networks, in a way that the origin and destination machines do not need to know where they are located).
In this scenario, you can only use a real phone number

Voice over IP: With Skype and a few other early services, appeared the possibility to make phone calls to classical phones (POTS) from a computer and also receive phone calls from those phones. In this case, the service provides a platform that sends/receives calls, and the call is routed via Internet to the destination (If implemented properly, it will only use internet in part of the path, and use other more efficient paths where possible).
In this scenario, you could use a real phone number.

Internet phone calls: This is the classic method of talking with other people, and what the Original poster was asking for: This was made popular with some Instant Messaging applications, like Messenger, and is also the usual way to make calls with Skype (when not having the paid service).
Since the main use was internet, and communications speeds weren't what they are today, lossy codecs are used, and latency might not be as good as possible.
In this scenario, you need to have the other user as a contact in the program

Mixtures of VoIP and Internet Phone: In the recent years, a nice solution for phone in enterprises has been a Digital switchboard, and having USB phones (like: http://www.polycom.com/products-services/v...ip-321-331.html ). The telephones are connected via USB to the PC, are seen as a soundcard by the PC, and the communication program uses it for transmitting audio and other advanced feature.
In this scenario, you can either use the contact, or a phone number. (Generally, as a contact you have an extension number, internal to the enterprise and the enterprise can have several lines to outside).


Footnote:
ADSL: The phone service provided usually with ADSL is not VoIP. ADSL technology allows reusing the signal used for classic telephone to also send internet service, but both services just share the cable, nothing more.
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phofman
post Jul 14 2013, 20:21
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IMO the key information is still missing - why do you need communication with so low latency. I would understand if you wanted to run networked computers doing audio work which requires low latency - for that specifically jackd + netjack were developed.
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slks
post Jul 15 2013, 09:29
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To put the "3 ms of latency" figure into perspective:

3ms = 0.003 seconds

Or, about 1/330th of a second.

As someone said before me, a cell phone might have 1/4 or 1/5 of a second latency, which is noticeable, but not bad.
1/50th of a second latency, not humanly perceptible.
1/330th of a second latency, not perceptible to the point of ridiculousness.

Unless you have some kind of superpower that lets you slow down time - in which case, my apologies.

Additionally, "I have extra bandwidth" has never been a valid reason to waste bandwidth. Eventually, you will have something that actually needs to use the extra bandwidth. Or, someone else in your neighborhood will have a need to use the extra bandwidth. When you waste bandwidth, it doesn't only waste your bandwidth, but everyone in your neighborhood's bandwidth. You are on the same network they are.


--------------------
http://www.last.fm/user/sls/
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zerowalker
post Jul 15 2013, 09:38
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QUOTE (slks @ Jul 15 2013, 10:29) *
To put the "3 ms of latency" figure into perspective:

3ms = 0.003 seconds

Or, about 1/330th of a second.

As someone said before me, a cell phone might have 1/4 or 1/5 of a second latency, which is noticeable, but not bad.
1/50th of a second latency, not humanly perceptible.
1/330th of a second latency, not perceptible to the point of ridiculousness.

Unless you have some kind of superpower that lets you slow down time - in which case, my apologies.

Additionally, "I have extra bandwidth" has never been a valid reason to waste bandwidth. Eventually, you will have something that actually needs to use the extra bandwidth. Or, someone else in your neighborhood will have a need to use the extra bandwidth. When you waste bandwidth, it doesn't only waste your bandwidth, but everyone in your neighborhood's bandwidth. You are on the same network they are.



Yes i know, the point is, i am not trying to waste, case itīs fun.
I got my own reasons for it. Itīs not like i want to run 100mbit down 24/7 just to make use of my internet.


But i am currently trying to make an application for this, which will take years at my programming skills.
Though i stumbled on something very interesting.

Webrtc, the latency there is extremely small, i am utterly impressed.
Sadly i donīt know hot to implement it myself or if itīs possible to control the bitrate etc, but for those who donīt know about, it may be worth checking out, quite fascinating.
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lozenge
post Jul 15 2013, 12:15
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You could use do something like pipe the audio output to netcat sending to an IP & port no.

CODE
< /dev/dsp | nc  remote.host.ip 4444

on the sending side, and


using netcat to listen on the local port redirected to the audio output.
CODE
nc -l 4444 > /dev/dsp

on the listening side.

You can add lame / sox / ffmpeg to do some encoding / decoding in between if required.

Sorry for the rushed answer, but have a play.

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phofman
post Jul 15 2013, 14:26
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QUOTE (lozenge @ Jul 15 2013, 13:15) *
You could use do something like pipe the audio output to netcat sending to an IP & port no.


It certainly works but IMO is not optimal in terms of latency. Piping introduces quite large latencies. Plus the alsa-oss emulation (the case for vast majority of linux distributions) goes through another layer of alsa-lib which most likely allocates its own buffer too.

Unfortunately in the long run it will produce over/underruns as two independent clock domains are involved.

In the unix world I would use jackd + netjack, a tool designed for the purpose of zero/low latency audio communication.
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