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138 lines
6.2 KiB
138 lines
6.2 KiB
The new Jitterbuffer in Asterisk
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Steve Kann
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The new jitterbuffer, PLC, and the IAX2-integration of the new jitterbuffer
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have been integrated into Asterisk. The jitterbuffer is generic and work is
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going on to implement it in SIP/RTP as well.
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Also, we've added a feature called "trunktimestamps", which adds individual
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timestamps to trunked frames within a trunk frame.
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Here's how to use this stuff:
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1) 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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2) 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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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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3) Trunktimestamps:
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-------------------
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To use this, both sides must be using Asterisk v1.2.
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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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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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4) 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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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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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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5) 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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"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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The stats shown are:
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* Jit: The jitter we have measured (milliseconds)
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* Del: The maximum delay imposed by the jitterbuffer (milliseconds)
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* Lost: The number of packets we've detected as lost.
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* %: The percentage of packets we've detected as lost recently.
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* Drop: The number of packets we've purposely dropped (to lower latency).
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* OOO: The number of packets we've received out-of-order
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* Kpkts: The number of packets we've received / 1000.
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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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1) 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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2) 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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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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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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3) 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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Reporting possible bugs
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-----------------------
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If you do find bad behaviors, here's the information that will help to diagnose this:
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1) Describe
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a) the source of the timestamps and frames: i.e. if they're coming from another chan_iax2 box,
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a bridged RTP-based channel, an IAX2 softphone, etc..
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b) The network between, in brief (i.e. the internet, a local lan, etc).
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c) What is the problem you're seeing.
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2) Take a look and see what iax2 show netstats is saying about the call, and if it makes sense.
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3) a tcpdump of the frames, (or, tethereal output from), so we can see the timestamps and delivery
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times of the frames you're receiving. You can make such a tcpdump with:
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tcpdump -s 2048 -w /tmp/example.dump udp and port 4569 [and host <other-end>]
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Report bugs in the Asterisk bugtracker, http://bugs.digium.com.
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Please read the bug guidelines before you post a bug.
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Have fun!
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-SteveK
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