[asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss

[Martin Joseph]

On 2007-03-24 01:53:16 -0700, Edoardo Serra 
<edoardo.serra at webrainstorm.it> said:

> Hi Francois,
> 
> f6hqz-m at hamwlan.net ha scritto:
>> Hi men,
>>  I have already encountered some issue like this with few switches (very
> 
>> known great brand)  which doesn't like VoIP traffic !
> 
> I also have switches of a very known great brand !!
> It was so strange to me that I didn't consider a network problem...
> 
>> Check by drectly connected the VoIP equipment - if you can - with 
>> temporary long Ethernet cables bypassing the tested switch to see what 
>> happens in this case.
> 
> I'd try to bypass the switch someway but every server neeeds to have 
> its own public ip address..
> I'll put an RTP proxy somewhere...
> 
>> You can also tell to "qualify" with a longer delay, but this could not 
>> help in case of regulary frames losses.
> 
> What about turning qualify off ?
> Do you think taht Asterisk is stopping RTP when it loose a qualify packet  ?
> Or is the RTP traffic itself that is lost by the switches ?

The fact that qualify fails means you have a network issue.  The same 
reason qualify fails (ie servers can't communicate) is the reason your 
users are experiencing quality issues in call.

turn off Qualify isn't going to fix anything IMO.  It's just going to 
hide it from you.

If the asterisk servers are all on your LAN then the network issue 
should be easily fixable.  If the Asterisk servers are at remote 
locations and are using public internet, you might have problems 
resolving this completely.

Marty