[asterisk-users] Two Connected Servers Sound Quailty

[Yossi Ben Hagai]

Hi Matt,

you didn't mention what type/bw of each site Internet connection, i suggest
that you try to split the scenario into smaller pieces:
- run long term pings between the server while you make a call and check for
packet loss.
- make internal calls between extensions on the same branch and verify that
both servers work okay (eliminating Internet connectivity)
- register a UA from one site to a server on the other site, make a call and
viceversa (eliminating a problem on one of the servers).
- check for speed/duplex setting on NIC and switch port.
- check if the sound quality issues are symmetric (does both sides
experience the sound cut or it only happens on a specific site).
- make sure you don't use G.711 as it consumes bw and from the codec
list you've mentioned has the lowest tolerance to packet loss.

Since the problems are intermittent my bet is that someone in the office is
have the p2p client work overtime or sending lots emails with funny
attachments


On 4/28/07, Matt Gardner <garddawg at gmail.com> wrote:
>
> Ok this is my first post and I will try to keep it short.
>
> I have searched everywhere and haven't found an answer to my question
>
> I have two Trixbox servers that are connected over the Internet via an
> IAX2 connection.  We are experiencing very poor sound quality.  I have tried
> many different codecs gsm, ilbc, g729, g711 and all seem to have the same
> problem. (All though g729 seems to work the best but still isn't reliable)
> The problems are intermittent sometimes the sound will cut out for 3-4
> seconds and other times the sound will just be loosing every other word, and
> other times it sounds just fine.
>
> Also, we have been using Skype over this same Internet connection and have
> very good sound quality with very few lost words.
>
> So here are my questions.
>
> First, is it a correct assumption to say that because Skype works well
> over this connection then I should be able to get asterisk to work over this
> connect.  I am hoping that Skype isn't "better" then asterisk in this area.
>
> If I should be able to get the same sound quality could you point me in
> the right direction on how to achieve this.  (I have tried messing with the
> jitterbuffer but haven't been able to find very good docs on how to utilize
> this functionality so about all I have done is set jitterbuffer=yes)
>
> Thanks in advance.
>
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