[Asterisk-Users] Static on inside end of conversation

[Jeff Busch]

Correct.

The issue is that most of the echo is between internal stations.  SIP ->
SIP.  

The users with the system using the sipura's don't report any echo when
calling outside the office or receiving a call.  

The users with the system using the audiocodes report an echo for the
first 1 - 2 seconds of a conversation using the pots lines (this must be
the echo cancellation process of the audiocodes) but then it clears up.
Likewise, they are experiencing echo when talking back and forth between
extensions.  I also forgot to mention that the users of the audiocodes
system report "static" on the line.

Jeff Busch 



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
"ManxPower" Wieling
Sent: Wednesday, December 07, 2005 11:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Static on inside end of conversation

The device that interfaces to the PSTN is the interface that must cancel
echo.  If I read your post correctly, that is the SAP-3000 and the
Audiocodes boxes in your case.

Jeff Busch wrote:
> Update on this... And it is still not solved.
> 
> This is actually fairly interesting.  I have two installations at a 
> construction company.  They are both running similar class machines (I

> was wrong in my initial post) they are:
> 
> System "A"
> 2.4 ghz Celeron
> 1 gb RAM
> IDE Drives
> Asterisk at Home 1.13 (Asterisk 1.0.9)
> An Audiocodes MP-108 to interface with PSTN
> 9 Polycom IP-500
> Phone system is on 16 port linksys switch.  All PC's are on separate 
> 16 port switch.  Both switches are plugged into our VPN firewall 
> (Netscreen
> 5xp)
> 
> 
> System "B"
> 2.4 ghz Celeron
> 1 gb RAM
> IDE Drives
> Asterisk at Home 2.1 (asterisk 1.2)
> 5 Sipura SPA-3000  to interface with PSTN
> 12 Polycom IP-500
> Phone and data on NEW Netgear 48 port Smart Switch.  Voice is on Vlan 
> 02 with QoS set to HIGH.  Data is on Vlan 01 with QoS set to LOW.  
> Switch is connected to Netscreen 5xp VPN firewall
> 
> 
> Polycoms are running the newest possible firmware for this phone:
> Bootrom 2.6.1 and SIP 1.5.2
> 
> 
> Everything is working fine on both systems, but BOTH systems are 
> experiencing echo on the IP side of the conversation.  Some phones are

> worse than others.
> 
> System A is experiencing echo both on internal Station to station 
> calls, and on outbound calls.
> System B is experiencing echo on internal station to station calls, 
> and no one has complained about echo on outbound calls yet.
> 
> The Polycoms are running G.711u as their 1st configured codec (G711a 
> as
> #2 and G.729AB as #3) and Asterisk is configured to disallow=all & 
> allow=ulaw.
> 
> I have searched and searched concerning echo and Polycom phones and 
> haven't found anything that seems to be relevant.  Any help would be 
> appreciated.
> 
> One more note, I am NOT running TDM4xxp cards on either of these 
> machines, so Zaptel information concerning echo wouldn't be relevant, 
> correct?
> 
> Thanks!
> 
> 
> Jeff Busch
> 
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike 
> McMullen
> Sent: Tuesday, November 29, 2005 6:26 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Static on inside end of conversation
> 
> 
> ----- Original Message -----
> From: "Jeff Busch" <Jeff.Busch at lewisbuilds.com>
> 
>> Hello,
>>
>> I am running the following configuration:
>>
>> 2.8ghz P4 with 1GB of RAM
>> Audiocodes MP-108 connected to 5 POTS lines Polycom IP-500 phones 
>> Asterisk at Home 1.3 (this is Asterisk 1.0.9)
>>
>> End users are complaining of an echo and static on the inside end 
>> (the
> 
>> internal side), but the outside end of the conversation doeesn't 
>> notice anything.
>>
>> Does anyone have any suggestions on troubleshooting / fixing this 
>> problem?
>>
> 
> Hi Jeff,
> 
> I recommend upgrading to Asterisk at home 2.0 which was just released. 
> It uses Asterisk 1.2 and a 2.6.9 based kernel which handles i/o and 
> interrupts much better.
> 
> While the link below discusses issues with digium cards, in general 
> the interrupts and IDE vs SATA drive discussions are of use no matter
what.
> 
> http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te
> 40
> 5p_noise.html
> 
> I have 2 Digium T400P cards connected to 8 POTS lines and 10 sipura
> spa-841 phones.
> 
> I went through two PCs we had (2.9 GHZ celeron, 2.1HGZ Athlon
> XP) both with IDE drives. I finally declared war on echo and the Rice 
> Krispies syndrome (Snap, Crackle, Pop) on the internal end of the 
> conversations.
> 
> I went and bought an ASUS P5LD2 motherboard with 1GB memory, 3.2GHz  
> P4 with hyperthreading and a 2MB cache, and a SATA drive. I installed 
> Asterisk at Home 2.0. After some minor magic to get the correct merlin 
> gigabit ethernet driver for CentOS 4.2, everything came up perfect.
> 
> This was my thanksgiving project and so far sound quality has been 
> perfect. No echo and no rice krispies.
> 
> I simulated network load on the system by copying multi-GBs of files 
> through the net from another server with scp while I called out and 
> back into the system on multiple lines. Even with the scp reporting 
> 9.5MB/sec, the phone sound quality was fantastic.
> 
> I then upped the ante by copying multi-gb files on the hard drive 
> which when viewing stats with top (hyperthreading shows as 2 CPUs and 
> you run the SMP kernel) both CPUs showed no idle time. IO- Wait state 
> never greater than 10%. Phone calls were still perfect with no echo or
noise.
> Out of the 10 vmails I left as part of the test, only one had
> 3 very faint pops in a 30 second message. They could have come from 
> the POTS line for all I know.
> 
> I ran extended phone conversations by calling the Asterisk system from

> our old phone system, picking up the extension on the called SIP phone

> and then playing Law & Order dvd episodes (lots of talking) and 
> placing the handset near to the speaker and taking the old phone 
> system handset and listening and talking back into it for 30 minutes
at a time.
> Necessity is indeed a mother.... ;-)
> 
> The calls were perfect. I'm amazed at how clear the dvd audio came 
> through.
> 
> I then reversed the process and played the dvd audio through the 
> Asterisk system handsets while listening and talking back through the 
> old phone handsets.
> 
> After 30 minutes the quality was still excellent.
> 
> 
> Hope you find some of this ramble useful.
> 
> Mike
>  
> 
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