OPUS > Yealink choppy audio

Asterisk Asterisk Endpoints
1

Yealink recently released OPUS capable firmware for some of their phones. I’m trying to get this to work but am having some final issues I need some help with.

My codecs.conf

[opus16v]
type=opus
packet_loss=20
complexity=2
signal=voice
application=voip
max_playback_rate=24000
bitrate=16000
cbr=no
fec=yes
dtx=no

Asterisk version is 13.14.0

I can place calls and everything is functional.

The problem is that the audio is choppy from asterisk>yealink. Not completely, but it has little breaks like g.711 with a bit of packet loss. yealink>asterisk is fine, if I connect up a g.711 call the yealink(opus)>asterisk>g.711 endpoint is clean.

PBX is remote, but 10ms away with no packet loss and jitter peaks at about 10ms. I run icmp pings across to try to track the gaps to any kind of loss but see no correlation. I capture a PCAP and there are no missing packets.

Any ideas what might be causing this? I’ve tried many different bitrates and complexity but they all behave the same. Evan a complexity=1 8kbps stream has this as does a compleixty=10 48kbps stream…

Thanks for any help.

2

I should add that if asterisk is just passing through opus (ie, both endpoints are opus) then it sounds great. Asterisk is definitely where the issue is.

3

If you just use the normal “opus” built in format what happens?

4

I’ll try now, I honestly didn’t know there was a built in [opus]… I dont have an [opus] entry in codecs.conf…

5

The codec_opus module automatically creates it. It’s there for most cases if you don’t need to tightly control things.

6

still does it. verified it’s just ‘opus’ in sip show channels.

7

I’m not able to reproduce this using a Digium D65 as an endpoint w/ that same opus16v codec in my codecs.conf. Local network; no loss or substantial jitter. Other endpoint is a D80. Tried w/ D80 using both G.722 and G.711 u-law. Audio towards Digium phone is good; audio from Digium phone is good. Echo app from Digium phone is fine. Playbacks of prompts (source file is slin44) are good. Asterisk-14 branch. Don’t have a 13.14 handy.

8

I can’t move to asterisk 14 yet, I could move up to 13.15 and see if that improves it.

9

Not really “the same” but there was recently an issue created for Choppy audio when using the open source opus codec with Snom handsets.

Now - the codec from Digium is completely different (I would presume) to the open source one - but Snom had to fix their handsets with an update - maybe Yealink went down a similar route?

10

The timestamp problem would impact both, as it is in the RTP layer and not the codec translation.

11

I was suspicious of the yealink implementation, or maybe some oddity between the yealink and asterisk implementations, but I’ve tested with Bria (android) , X-Lite, and Grandstream HT802 and they all do it. X-lite does it a tiny bit less, or rather it seems to have a shorter and less noticeable gap than the others but it’s still there.

12

I don’t have Bria, X-Lite or a Grandstream HT802. I do have Linphone and Jitsi. They’re both fine for me. Source device is still a Digium D80.

13

which version of asterisk are you running?

14

Asterisk GIT-14-0cec000

15

13.14.0 here and I can’t go to 14. I’m going to do 13.15 after hours and see if that changes anything.