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	- Many uses of the astlisting environment around verbatim text to ensure that it gets properly formatted and doesn't run off the page. - Update some things that have been deprecated. - Add escaping as needed - and more ... (closes issue #10978) Reported by: IgorG Patches: texdoc-85542-1.patch uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
		
			
				
	
	
		
			99 lines
		
	
	
		
			4.9 KiB
		
	
	
	
		
			TeX
		
	
	
	
	
	
			
		
		
	
	
			99 lines
		
	
	
		
			4.9 KiB
		
	
	
	
		
			TeX
		
	
	
	
	
	
| \subsubsection{The new jitterbuffer}
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| 
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| You must add "jitterbuffer=yes" to either the [general] part of
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| iax.conf, or to a peer or a user.  (just like the old jitterbuffer).
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| Also, you can set "maxjitterbuffer=n", which puts a hard-limit on the size of the
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| jitterbuffer of "n milliseconds".  It is not necessary to have the new jitterbuffer
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| on both sides of a call; it works on the receive side only.
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| 
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| \subsubsection{PLC}
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| 
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| The new jitterbuffer detects packet loss.  PLC is done to try to recreate these
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| lost packets in the codec decoding stage, as the encoded audio is translated to slinear.
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| PLC is also used to mask jitterbuffer growth.
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| 
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| This facility is enabled by default in iLBC and speex, as it has no additional cost.
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| This facility can be enabled in adpcm, alaw, g726, gsm, lpc10, and ulaw by setting
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| genericplc =$>$ true in the [plc] section of codecs.conf.
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| 
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| \subsubsection{Trunktimestamps}
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| 
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| To use this, both sides must be using Asterisk v1.2 or later.
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| Setting "trunktimestamps=yes" in iax.conf will cause your box to send 16-bit timestamps
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| for each trunked frame inside of a trunk frame. This will enable you to use jitterbuffer
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| for an IAX2 trunk, something that was not possible in the old architecture.
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| 
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| The other side must also support this functionality, or else, well, bad things will happen.
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| If you don't use trunktimestamps, there's lots of ways the jitterbuffer can get confused because
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| timestamps aren't necessarily sent through the trunk correctly.
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| 
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| \subsubsection{Communication with Asterisk v1.0.x systems}
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| 
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| You can set up communication with v1.0.x systems with the new jitterbuffer, but
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| you can't use trunks with trunktimestamps in this communication.
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| 
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| If you are connecting to an Asterisk server with earlier versions of the software (1.0.x),
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| do not enable both jitterbuffer and trunking for the involved peers/users
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| in order to be able  to communicate. Earlier systems will not support trunktimestamps.
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| 
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| You may also compile chan\_iax2.c without the new jitterbuffer, enabling the old
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| backwards compatible architecture. Look in the source code for instructions.
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| 
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| 
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| \subsubsection{Testing and monitoring}
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| 
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| You can test the effectiveness of PLC and the new jitterbuffer's detection of loss by using
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| the new CLI command "iax2 test losspct $<$n$>$".  This will simulate n percent packet loss
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| coming \_in\_ to chan\_iax2. You should find that with PLC and the new JB, 10 percent packet
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| loss should lead to just a tiny amount of distortion, while without PLC, it would lead to
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| silent gaps in your audio.
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| 
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| "iax2 show netstats" shows you statistics for each iax2 call you have up.
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| The columns are "RTT" which is the round-trip time for the last PING, and then a bunch of s
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| tats for both the local side (what you're receiving), and the remote side (what the other
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| end is telling us they are seeing).  The remote stats may not be complete if the remote
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| end isn't using the new jitterbuffer.
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| 
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| The stats shown are:
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| \begin{itemize}
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| \item Jit: The jitter we have measured (milliseconds)
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| \item Del: The maximum delay imposed by the jitterbuffer (milliseconds)
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| \item Lost: The number of packets we've detected as lost.
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| \item \%: The percentage of packets we've detected as lost recently.
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| \item Drop: The number of packets we've purposely dropped (to lower latency).
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| \item OOO: The number of packets we've received out-of-order
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| \item Kpkts: The number of packets we've received / 1000.
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| \end{itemize}
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| 
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| \subsubsection{Reporting problems}
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| 
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| There's a couple of things that can make calls sound bad using the jitterbuffer:
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| 
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| \begin{enumerate}
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| \item The JB and PLC can make your calls sound better, but they can't fix everything.
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| If you lost 10 frames in a row, it can't possibly fix that.  It really can't help much
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| more than one or two consecutive frames.
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| 
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| \item Bad timestamps:  If whatever is generating timestamps to be sent to you generates
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| nonsensical timestamps, it can confuse the jitterbuffer.  In particular, discontinuities
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| in timestamps will really upset it:  Things like timestamps sequences which go 0, 20, 40,
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| 60, 80,  34000, 34020, 34040, 34060...   It's going to think you've got about 34 seconds
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| of jitter in this case, etc..
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| The right solution to this is to find out what's causing the sender to send us such nonsense,
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| and fix that.  But we should also figure out how to make the receiver more robust in
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| cases like this.
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| 
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| chan\_iax2 will actually help fix this a bit if it's more than 3 seconds or so, but at
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| some point we should try to think of a better way to detect this kind of thing and
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| resynchronize.
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| 
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| Different clock rates are handled very gracefully though; it will actually deal with a
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| sender sending 20\% faster or slower than you expect just fine.
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| 
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| \item Really strange network delays:  If your network "pauses" for like 5 seconds, and then
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| when it restarts, you are sent some packets that are 5 seconds old, we are going to see
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| that as a lot of jitter.   We already throw away up to the worst 20 frames like this,
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| though, and the "maxjitterbuffer" parameter should put a limit on what we do in this case.
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| 
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| \end{enumerate}
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