[asterisk-bugs] [JIRA] (ASTERISK-21872) RFC2833 DTMF Events received at a very fast interval, ~25ms, results in high CPU and when running in -p results in RTP packet sending delays
hristo (JIRA)
noreply at issues.asterisk.org
Wed Jun 26 11:34:04 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-21872?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=207519#comment-207519 ]
hristo commented on ASTERISK-21872:
-----------------------------------
{quote}
1. Can you have your generators/devices send DTMFs slower?
{quote}
Certainly, as far as testing is concerned. However, I am able to reproduce this with a single phone (tested with SNOM 370, SNOM 320) and a single finger. The only reason I use a call generator is to give my fingers some time to rest. I don't think that I am able to hit the dialpad in 25ms intervals, but I will verify with a packet capture. My bigger problem is that this also happens during completely normal usage, which is why I originally opened the ticket.
Take for example a conference bridge service - the participants tend to connect all at the same time. Let's take only a single 20-participant conference, each participant having to dial a 6 digit PIN number and almost all usually trying to connect at about the time the conference starts. It is not unlikely that from the total of 120 DTMFs Asterisk will have to process some of them in bursts (possibly with 25ms intervals). Also, the CPU is probably under 10-20% during normal usage, when no DTMFs are hitting the server, so it will really be a waste of resource to do the capacity planning based on the DTMFs performance.
BTW, the conference bridge service runs on a dedicated system, so Asterisk is really only forwarding the RTP/Events stream in this case.
{quote}
2. Do you need to run Asterisk with -p
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I am considering removing it. I don't think it is really needed in this particular case, but it was set by someone else.It took me some time to discover it myself. This will hopefully resolve the RTP delay problem, only leaving the high CPU usage problem, which is worrisome by itself.
{quote}
Let us know if you can reproduce the overly high CPU consumption when sending DTMF with a interval of 50ms or above with a few channels.
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Will try to test this exact scenario tomorrow and will report back.
> RFC2833 DTMF Events received at a very fast interval,~25ms, results in high CPU and when running in -p results in RTP packet sending delays
> -------------------------------------------------------------------------------------------------------------------------------------------
>
> Key: ASTERISK-21872
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-21872
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Core/General
> Affects Versions: 1.8.17.0, 1.8.19.1, 1.8.20.0, 1.8.22.0
> Environment: Debian 6.0 64-bit
> Reporter: hristo
> Assignee: hristo
> Severity: Minor
> Attachments: 2-calls-one-sending-many-dtmfs-asterisk-debug.txt, forward-stream-first-call-after-asterisk.pcap.txt, forward-stream-first-call-before-asterisk.pcap.txt, sample-config.diff, vmstat.txt
>
>
> If I send several DTMFs to Asterisk, one after the other, fast enough, it blocks other voice RTP packets for as long as several hundred milliseconds. This seems to affects *all* RTP streams on a server.
> I can say for sure, that Asterisk is not dropping the RTP packets, because after a while it sends all of them at once. It seems as if they are being held by something, while the DTMFs are being processed/forwarded.
> This only occurs in non Packet2Packet mode.
> Originally I've seen the problem when several people were connecting to a conference at about the same time and were entering the PIN numbers at about the same time, therefore producing a lot of DTMFs. The conference runs on a dedicate hardware und is unrelated. Asterisk just sits in the middle and bridges the calls. I have managed to reproduce this with only two calls with as little as 10-15 DTMFs, provided they are send fast enough.
> Attaching is a debug console log from the following call scenario. In this case both calls were genereted from a dedicated server and terminated on another dedicated server.
> Call 1:
> A (IP 1.1.1.1) dials 1000 --> Asterisk (IP 2.2.2.2) ---> B (IP 3.3.3.3)
> Call 2:
> A' (IP 1.1.1.1) dials 2000 --> Asterisk (IP 2.2.2.2) ---> B' (IP 3.3.3.3)
> Both calls are active at this point. A' on Call 2 starts sending DTMFs (in this case 40 of them). As a result RTP packets from Call *1* in both directions are delayed by 150-160 ms and are being sent in bursts.
> In the logs I often see:
> res_timing_timerfd.c:225 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead
> and the CPU is close to 100% (caused by the asterisk process). As soon as all DTMFs are sent, the RTP streams return back to normal with asterisk sending one packet every 20 ms on average.
> Attached is also a filtered packet capture that shows only the forward RTP stream on Call 1 from A -> Asterisk and from Asterisk -> B. "Time" represents the delta from the previos packet. Under normal conditions this should be close to 0.020 s (or 20 ms).
> One example of the problem can be seen at line 1235 in 'forward-stream-first-call-after-asterisk.pcap.txt'. The packet there has been held for ~160 ms, then sent together with the next 7 packets all at once.
> The RTP packets from the corresponding call leg (before asterisk) start at line 1244 in 'forward-stream-first-call-before-asterisk.pcap.txt" and are all equally spaced at about 20 ms.
> There are many such examples - simply search for 0.000 (deltas which are less than 1 ms) to identify groups of packets that are sent together. The same problem is present in the backward stream too (not attached).
> How to reproduce - add the following to the dialplan:
> exten => _X.,n,Dial(SIP/B at 3.3.3.3,,t)
> The 't' option is important, because it effectively disables the Packet2Packet mode. Connect 2 calls (2 sets of telephones) and start dialing DTMFs as fast as you can on one of them (or simply generate 2 calls and send the DTMFs as I did). This will disrupt the call between the other set of phones if done fast enough.
> I have tested this on 3 servers (2 physical and one virtual). All of them were running the same OS (Debian 6), so this may end up being an OS or res_timing_timerfd problem after all, but I really cannot test it on a different distribution.
> I tested with the following versions and was able to reproduce the problem with all of them:
> 1.8.22.0
> 1.8.20.0
> 1.8.19.1
> 1.8.17.0
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