[asterisk-bugs] [JIRA] (ASTERISK-25629) [patch] Native Packet-Loss Concealment (PLC)

Wed Oct 5 06:36:01 CDT 2016

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25629?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232525#comment-232525 ] 

Steve Davies . commented on ASTERISK-25629:
-------------------------------------------

Hi,

Comment from a user - I applied this patch to my 13.10.0 system and it really didn't work well for me.

One case was a call which was being transcoded alaw -> (slin) -> g729.

The user on the g729 side called up an inband transfer (# on our system).  At that point :

{code:title=full log|borderStyle=solid}
[Oct  4 12:24:33] DTMF[28899][C-0000d7a3] channel.c: DTMF begin '#' received on SIP/41.221.230.19:5060-0001bd56
[Oct  4 12:24:33] DTMF[28899][C-0000d7a3] channel.c: DTMF begin passthrough '#' on SIP/41.221.230.19:5060-0001bd56
[Oct  4 12:24:33] DTMF[28899][C-0000d7a3] channel.c: DTMF end '#' received on SIP/41.221.230.19:5060-0001bd56, duration 60 ms
[Oct  4 12:24:33] DTMF[28899][C-0000d7a3] channel.c: DTMF end accepted with begin '#' on SIP/41.221.230.19:5060-0001bd56
[Oct  4 12:24:33] DTMF[28899][C-0000d7a3] channel.c: DTMF end '#' detected to have actual duration 61 on the wire, emulation will be triggered on SIP/41.221.230.19:5060-0001bd56
[Oct  4 12:24:33] DTMF[28899][C-0000d7a3] channel.c: DTMF end '#' has duration 61 but want minimum 80, emulating on SIP/41.221.230.19:5060-0001bd56
[Oct  4 12:24:33] DTMF[28899][C-0000d7a3] channel.c: DTMF end emulation of '#' queued on SIP/41.221.230.19:5060-0001bd56
[Oct  4 12:24:33] VERBOSE[28899][C-0000d7a3] file.c: <SIP/41.221.230.19:5060-0001bd56> Playing 'pbx-transfer.g729' (language 'en')
[Oct  4 12:24:33] VERBOSE[28544][C-0000d7a3] res_musiconhold.c: Started music on hold, class '2521', on channel 'Local/7278200 at enswitch-phone-000134c9;2'
[Oct  4 12:24:33] NOTICE[28509][C-0000d7a3] translate.c: 18772 lost frame(s) 0/46762 (g729 at 8000)->(slin at 8000)->(alaw at 8000)
[Oct  4 12:24:33] VERBOSE[28509][C-0000d7a3] codec_g72x.c:     -- G.729 PLC
..
{code}

At that point my log has 18772 "G.729 PLC" messages all in the same second.

As far as I can tell the logic decided to ask g729 codec to construct 18772 "plc" frames.  That seems crazy and the result is not pretty.

Regards,
Steve

> [patch] Native Packet-Loss Concealment (PLC)
> --------------------------------------------
>
>                 Key: ASTERISK-25629
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25629
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/codec_ilbc, Codecs/codec_speex, Codecs/General
>    Affects Versions: 11.22.0, 13.9.1
>            Reporter: Alexander Traud
>            Assignee: Alexander Traud
>            Severity: Minor
>         Attachments: native_plc.patch
>
>
> In VoIP/SIP, the RTP media is transferred not via reliable TCP but [UDP|https://en.wikipedia.org/wiki/User_Datagram_Protocol#Comparison_of_UDP_and_TCP]. This makes sure the data arrives as fast as possible. However, because of the use of UDP, packets might never arrive or take another route via the Internet. Therefore, one packet might be faster than previous packets. Or stated differently: Packets might arrive late. In both cases, those packets are lost because they cannot be used to re-build the media stream.
> Therefore, VoIP specific audio-codecs like iLBC, Speex, and SILK/Opus are able to conceal lost packets. This is called Packet-Loss Concealment (PLC). In this case, it is called native PLC, because no additional source code must be written, but the underlying library supports this already. Additionally, Asterisk offers generic PLC while writing Signed-Linear.
> To conceal packet-loss, the codec library must be aware that a packet got lost. Furthermore because RTP packets inter-depent, late packets must not be forwarded to the library not to confuse its state. If a late packet arrives, the library would not know that it has to discard it, because the library does not know its RTP sequence number. The VoIP application – in our case Asterisk – has to discard late packets and indicate lost packets to the library. This is true for iLBC, Speex, SILK, and Opus Codec.
> Long story short: In contrast to [the documentation|https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5243109], native PLC does not happen. Without or with a jitter buffer.
> All Long-Term-Support releases (including Asterisk 1.8.32.3, Asterisk 1.4.44, and Asterisk 1.2.40) were re-tested with a simple setup: Two Wi-Fi access points bridged via Ethernet to the same DHCP server, using the same SSID, and just WPA2-Personal but sending on different 2,4 GHz channels (Wi-Fi Roaming). Plus a Wi-Fi enabled VoIP/SIP client, moving around those access point. Whenever the Wi-Fi client changes the access point, it has to authenticate again. That creates packet loss of about one to several dozen packets. In those Asterisk releases, their iLBC transcoding module was passed. The backtrace is
> main/channel.c:ast_read(.)
>   main/translate.c:ast_translate(.)
>     main/translate.c:framein(.)
>       codecs/codec_ilbc.c:framein(.)
> but native PLC was not done, because there was no indication of lost packets. Furthermore, late packets were not dropped but forwarded as usual packets. Those packets with unexpected RTP sequence numbers were double-checked via Wireshark » Menu » Telephony » RTP » Show All Streams » Analyse » Next non-OK.

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