[asterisk-bugs] [JIRA] (ASTERISK-21872) high CPU usage ~15 seconds into call if rtpkeepalive set on channels when Asterisk is in a generic bridge and passing RFC2833 DTMF

[hristo (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21872?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=207919#comment-207919 ] 

hristo commented on ASTERISK-21872:
-----------------------------------

I did a copy/paste of the peer and all its settings from a working configuration while truing to apply the minimum set of changes to the sample configs in order to reproduce the problem. That's how the rtpkeepalive setting ended up in my test configs.

I did another round of testing and at least for me the rtpkeepalive setting makes no difference. I see the exact same problem without the rtpkeepalive setting too. I tested both on Ubuntu and Debian by removing the setting from the peer and also verified via "sip show settings" that it's disabled globally.

BTW, not sure if it is at all related, but I can see that the "rtpkeepalive" RTP packets according to wireshark always have the PCMU codec (regardless of the call negotiated codec). Also, they seem to be sent alongside the "normal" RTP packets. I thought that the "rtpkeepalive" packets were supposed to be sent only when there is no "normal" RTP activity - for example during hold and no configured MOH, but I never expected to see them as extra packets in an active RTP stream (which really makes no sense). However, this is a completely different problem. I am only mentioning it here for completeness and just in case it happens to be somehow related to the current ticket.
                
> high CPU usage ~15 seconds into call if rtpkeepalive set on channels when Asterisk is in a generic bridge and passing RFC2833 DTMF
> ----------------------------------------------------------------------------------------------------------------------------------
>
>                 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: SVN, 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, full.txt, sample-config.diff, trafficdump.pcap, 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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