[asterisk-bugs] [JIRA] (ASTERISK-21872) RFC2833 DTMF Events cause RTP packet delays and high CPU

Wed Jun 19 16:32:03 CDT 2013

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21872?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=207349#comment-207349 ] 

Rusty Newton edited comment on ASTERISK-21872 at 6/19/13 4:30 PM:
------------------------------------------------------------------

I wasn't able to reproduce this on SVN-branch-1.8-r391778 or 11.4.0. I followed your guidance in the description, but no luck.  (or good luck?)  I can bring up two simple SIP to SIP calls, media through Asterisk, and on one call send across 40-60 DTMF digits (2833) with 50ms interval and 250ms duration and it doesn't block or slow any RTP on the other call. Also attempting something similar by hand, but of course with a higher interval since my fingers can't tap at 50ms intervals ends up with the same results.

Asterisk was using res_timing_timerfd.so, but I don't think Asterisk is using a timing interface in this case.

You'll have to provide more detail on how to reproduce (perhaps sip.conf config for the endpoints) and any other details on configuration that may be relevant.

If you are doing a lot of other things on this system then I would recommend building Asterisk from default configs on another system for a very simple test scenario and trying to reproduce it there. If you can't, then compare between the two and slowly add pieces to the test system until you find out what triggers it.

This is one we'll have to reproduce to move forward. I'll leave this open for a few weeks to see if you can provide additional detail that would help.

If you do get a method for sure-fire reproduction in a clean asterisk install, then you may also want to send along full SIP/RTP pcaps that we can view with wireshark.

      was (Author: rnewton):
    I wasn't able to reproduce this on SVN-branch-1.8-r391778 or 11.4.0. I followed your guidance in the description, but no luck.  (or good luck?)  I can bring up two simple SIP to SIP calls, media through Asterisk, and on one call send across 40-60 DTMF digits (2833) with 50ms interval and 250ms duration and it doesn't block or slow any RTP on the other call. Also attempting something similar by hand, but of course with a higher interval since my fingers can't tap at 50ms intervals ends up with the same results.

Asterisk was using res_timing_timerfd.so, but I don't think Asterisk is using a timing interface in this case.

You'll have to provide more detail on how to reproduce (perhaps sip.conf config for the endpoints) and any other details on configuration that may be relevant.

If you are doing a lot of other things on this system then I would recommend building Asterisk from default configs on another system for a very simple test scenario and trying to reproduce it there. If you can't, then compare between the two and slowly add pieces to the test system until you find out what triggers it.

This is one we'll have to reproduce to move forward. I'll leave this open for a few weeks to see if you can provide additional detail that would help.

> RFC2833 DTMF Events cause RTP packet delays and high CPU
> --------------------------------------------------------
>
>                 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
>            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
>
>
> 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

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira