Lost frame(s) from slin to alaw

FreePBX Providers
1

Hello.

Using Freepbx 15.0.29 / Asterisk 16.21.1

Checking logs into my Freepbx, I can see that there are lost frames

    -- Channel PJSIP/1009-00000001 joined 'simple_bridge' basic-bridge <9717e7e6-f720-46f1-909e-cc9f9cdef9ec>
    -- Channel Local/600@from-queue-00000000;2 joined 'simple_bridge' basic-bridge <9717e7e6-f720-46f1-909e-cc9f9cdef9ec>
    -- Local/600@from-queue-00000000;1 answered PJSIP/OVH-SIP-00000000
    -- <Local/600@from-queue-00000000;1> Playing 'custom/Aslog.slin' (language 'fr')
       > 0xaac666e0 -- Strict RTP switching to RTP target address 192.168.20.92:3000 as source
[2023-03-03 09:18:31] NOTICE[4496][C-00000001]: translate.c:603 ast_translate: 4640 lost frame(s) 4641/0 (slin@8000)->(alaw@8000)
    -- Stopped music on hold on PJSIP/OVH-SIP-00000000
    -- Channel Local/600@from-queue-00000000;1 joined 'simple_bridge' basic-bridge <4532078c-1a34-42ce-aa99-f15119fa39c4>
    -- Channel PJSIP/OVH-SIP-00000000 joined 'simple_bridge' basic-bridge <4532078c-1a34-42ce-aa99-f15119fa39c4>
       > 0xaac666e0 -- Strict RTP learning complete - Locking on source address 192.168.20.92:3000

The line is

[2023-03-03 09:18:31] NOTICE[4496][C-00000001]: translate.c:603 ast_translate: 4640 lost frame(s) 4641/0 (slin@8000)->(alaw@8000)

If I check the stats…

raspbx4*CLI> pjsip show channelstats

                                             ...........Receive......... .........Transmit..........
 BridgeId ChannelId ........ UpTime.. Codec.   Count    Lost Pct  Jitter   Count    Lost Pct  Jitter RTT....
 ===========================================================================================================

 4532078c OVH-SIP-00000000   00:00:21 alaw     1029       0    0   0.000   1031       0    0   0.000   0.000
 9717e7e6 1009-00000001      00:00:20 alaw      783       0    0   0.000    793       0    0   0.000   0.012

In this particular case, the phone link was good.

Why are there lost frames ?

Thanks in advance.

2

Furthermore, this message seems to be associated with chan_dongle, which is third party code, and, to the best of my knowledge, no longer supported.

Asterisk 16 no longer receives bug fixes.

3

Hum… I tested this in the past to play with 2G voice calls.
Interesting…

But strange that this comes here while I do not use it anymore.

Anyway, I’m building new system based on old XIVO hardware where I installed last Freepbx version.
Will not have chan_dongle anymore.

Many thanks for your help.

Regards,
Laurent.

4

It is possible that it isn’t just chan_dongle, although that is the context in which others have reported it. I think it has to be RPi though, given that the message isn’t in the official code.

5

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