* You are viewing Posts Tagged ‘voice quality’

Obtaining High Voice Quality In Dahdi Channel

Hello;

What is the best method to let the voice quality through Dahdi channels to be clear and no echo? Is it the wanpipe or it is working only with sangoma?

Regards Bilal

Digium IP Phones D40

Hi All;

Any one used Digium IP Phones D40?

I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium.

Regards
Bilal

Transcoding degradation G711iLBC

On 04/15/2012 07:26 PM, Patrick Lists wrote:
> On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
>> Is it a good idea to use asterisk transcoding from G711 to iLBC or
>> should I find out any other solution not involving transcoding (f.e.
>> using G.729 that is supported in both sides). I’m worried about voice
>> quality and trying to avoid paying for G.729 licensing.
>>
>> Anybody with experience or quantitative measurements of the voice
>> quality degradation in that scenario?
>
> The term that may interest you is “Mean Opinion Score” and iLBC is
> quite good. See http://en.wikipedia.org/wiki/Mean_opinion_score
There’s lies, damn lies and mean opinion scores. The chart on that
wikipedia page is mostly for humour value.

Regards,
Steve

Transcoding degradation G711iLBC

On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
> Is it a good idea to use asterisk transcoding from G711 to iLBC or
> should I find out any other solution not involving transcoding (f.e.
> using G.729 that is supported in both sides). I’m worried about voice
> quality and trying to avoid paying for G.729 licensing.
>
> Anybody with experience or quantitative measurements of the voice
> quality degradation in that scenario?

The term that may interest you is “Mean Opinion Score” and iLBC is quite
good. See http://en.wikipedia.org/wiki/Mean_opinion_score

Regards,
Patrick

Transcoding degradation G711iLBC

Is it a good idea to use asterisk transcoding from G711 to iLBC or should I find out any other solution not involving transcoding (f.e. using G.729 that is supported in both sides). I’m worried about voice quality and trying to avoid paying for G.729 licensing.

Anybody with experience or quantitative measurements of the voice quality degradation in that scenario?

Regards,
G

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SIP jitter and packlost channel variables

Hi,

A client of ours get lots of problem with there voice quality when the do a lot SIP calls.
In a application I log the rtpqos audio jitter an lost packets. (see Below)

Does anybody know what the numbers mean?
If I look at a sample of the channel variables, I see the following number.
local_lostpackets = 7706
local_jitter = 2
local_maxjitter = 11
local_minjitter = 0
..
..
remote_lostpackets = 0
remote_jitter = 0
remote_maxjitter = 70000
remote_minjitter = 14000
..
..

The only thing I see is this: http://www.voip-info.org/wiki/view/Asterisk+func+channel

Regards,

Arjan Kroon
Mobillion BV

exten => s,n,Set(A_SIP_DATA=${CHANNEL(rtpqos,audio,local_lostpackets)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_jitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_maxjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_minjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_normdevjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_stdevjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_lostpackets)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_jitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_maxjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_minjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_normdevjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_stdevjitter)})