[asterisk-dev] Opus and VP8

Wed Jun 5 10:53:17 CDT 2013

Hi Andrea,

2013/6/5 Andrea Suisani <sickpig at opinioni.net>

> On 06/05/2013 03:07 PM, Andrea Suisani wrote:
>
>> On 06/05/2013 12:45 PM, Lorenzo Miniero wrote:
>>
>>> 2013/6/5 Andrea Suisani <sickpig at opinioni.net <mailto:
>>> sickpig at opinioni.net>>
>>>
>>
>> [cut]
>>
>>      as I already said this fix solve the issue we had.
>>>
>>>     the tests we made are related to this configuration:
>>>
>>>              (sip+opus)             (iax/sip+g729)
>>>     WebRTC ----------->  Asterisk -----------------> pstn/mobile
>>>                (A)                      (B)
>>>
>>>     I know that there is no ad-hoc Opus configuration yet,
>>>     but I'm trying to measure the bandwidth of the call opus leg (A).
>>>
>>>     I've used iftop to get a rough measure, catching only udp
>>>     packets. the iftop instance was executed on the same host
>>>     where asterisk is running catching packets only on the net
>>>     interface to which asterisk is bind to.
>>>
>>>     While using opus the A leg bandwidth is something like 50/55 kb/s,
>>>     from the webrtc client to the asterisk, while something along
>>>     100kb/s from asterisk to client. is it normal?
>>>
>>>
>>> Opus is conceived to be able to transparently switch rate during a call.
>>> In the code I made sure that
>>> the encoding would take note of the peer sampling rate, in order to cap
>>> it in Opus as well. This means that,
>>>
>>  > if you're talking to a GSM/ulaw/g729 peer, Opus itself should be
>> "capped" at 8kHz as
>>
>>> there would be no need for a higher sampling rate to transport the audio.
>>>
>>
> [cut explanation of a maybe false theory]
>
> scratch what I said before.
>
> I've reproduced the behavior (use 48kHz instead of 8kHz) with a plain and
> simple
> webrtc client (a local copy of http://tryit.jssip.net/) and I've tried to
> place
> a call without using AMI Originate through a simple dialplan Dial and I
> got the same
> result, this is the console log
>
>
The same scenario, in my setup, works fine: a browser calling a softphone
using a narrowband codec (e.g., u-Law) is capped to 8kHz, opus<->8000.

I guess the only difference between our scenarios (except for the several
MACROS that are not in my test extension) is the protocol: in my
extensions, only SIP is involved, and not IAX2. This may be what is causing
the issue, as recently someone posted a similar problem on github:

https://github.com/meetecho/asterisk-opus/issues/1#issuecomment-18926261

Unfortunately I'm unfamiliar with IAX, so I don't know how codecs are
negotiated and the related translation paths put in place for a call.

Lorenzo

> suino*CLI> opus set debug huge
> Opus debugging huge
>   == Using SIP RTP CoS mark 5
> [Jun  5 17:17:56] WARNING[31818][C-00000000]: chan_sip.c:11101
> process_sdp_a_audio: Got Opus minptime=10
>     -- Executing [93XX5XXXXX at iaxtest:1] Set("SIP/1060-00000000",
> "__session=") in new stack
>     -- Executing [93XX5XXXXX at iaxtest:2] Set("SIP/1060-00000000",
> "__numero=3XX5XXXXX") in new stack
>     -- Executing [93XX5XXXXX at iaxtest:3] Set("SIP/1060-00000000",
> "__prefix=9") in new stack
>     -- Executing [93XX5XXXXX at iaxtest:4] Macro("SIP/1060-00000000",
> "choose-provider,9") in new stack
>     -- Executing [s at macro-choose-provider:1] NoOp("SIP/1060-00000000",
> "9") in new stack
>     -- Executing [s at macro-choose-provider:2] Set("SIP/1060-00000000",
> "GLOBAL(PROVIDER)=IAX2/**myprovider") in new stack
>   == Setting global variable 'PROVIDER' to 'IAX2/myprovider'
>     -- Executing [s at macro-choose-provider:3] Set("SIP/1060-00000000",
> "GLOBAL(PROVIDER)=IAX2/**myprovider") in new stack
>   == Setting global variable 'PROVIDER' to 'IAX2/myprovider'
>     -- Executing [s at macro-choose-provider:4] Set("SIP/1060-00000000",
> "GLOBAL(PROVIDER)=IAX2/**myprovider") in new stack
>   == Setting global variable 'PROVIDER' to 'IAX2/myprovider'
>     -- Executing [s at macro-choose-provider:5] Set("SIP/1060-00000000",
> "GLOBAL(PROV_NOCHAN)=**myprovider") in new stack
>   == Setting global variable 'PROV_NOCHAN' to 'myprovider'
>     -- Executing [s at macro-choose-provider:6] Set("SIP/1060-00000000",
> "GLOBAL(PROV_CHAN)=IAX2") in new stack
>   == Setting global variable 'PROV_CHAN' to 'IAX2'
>     -- Executing [93XX5XXXXX at iaxtest:5] NoOp("SIP/1060-00000000",
> "IAX2/myprovider") in new stack
>     -- Executing [93XX5XXXXX at iaxtest:6] NoOp("SIP/1060-00000000",
> "myprovider") in new stack
>     -- Executing [93XX5XXXXX at iaxtest:7] NoOp("SIP/1060-00000000", "IAX2")
> in new stack
>     -- Executing [93XX5XXXXX at iaxtest:8] GotoIf("SIP/1060-00000000",
> "1?diax:dsip") in new stack
>     -- Goto (iaxtest,93XX5XXXXX,9)
>     -- Executing [93XX5XXXXX at iaxtest:9] GotoIf("SIP/1060-00000000",
> "0?backupadsl:dial") in new stack
>     -- Goto (iaxtest,93XX5XXXXX,11)
>     -- Executing [93XX5XXXXX at iaxtest:11] SendText("SIP/1060-00000000",
> "VOIP") in new stack
>     -- Executing [93XX5XXXXX at iaxtest:12] Set("SIP/1060-00000000",
> "CHANNEL(userfield)=") in new stack
>     -- Executing [93XX5XXXXX at iaxtest:13] Set("SIP/1060-00000000",
> "DYNAMIC_FEATURES=testfeature#**listensergiu#listensergiu2") in new stack
>     -- Executing [93XX5XXXXX at iaxtest:14] Dial("SIP/1060-00000000",
> "IAX2/myprovider/3XX5XXXXX,22)**") in new stack
>     -- Called IAX2/myprovider/3XX5XXXXX
> [Opus] Created encoder #3 (48000->opus)
> [Opus] Created decoder #6 (opus->48000)
>     -- Call accepted by MY.PRO.VIDER.IP (format g729)
>
>     -- Format for call is (g729)
>     -- IAX2/myprovider-2277 is making progress passing it to
> SIP/1060-00000000
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 220 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 119 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 119 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 125 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 123 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 124 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 123 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 185 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 119 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 118 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 120 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 120 bytes
> [Opus] [Encoder #3 (48000)] 960 samples, 1920 bytes
> [Opus] [Encoder #3 (48000)]   >> Got 960 samples, 120 bytes
>
>
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