asterisk users - Oct 2013 - Tired of dropouts and garbled phone calls - where to go next?

All,

The users in our organization are well, quite frankly, sick of phone
service that is being provided.  The choppy phone calls, and drop outs are
detrimental to our sales force.

I've tried about everything I can think of.

Moved the asterisk server from VM machine to dedicated machine

More than enough bandwidth

Setting 802.1p = 7

Set Dedicated voice traffic 35% of bandwidth.

Not sure what option would be the best


Put analog lines in the conference room to avoid the dropouts - leave the
sip lines in place for day to day use

Hire a consultant

Ditch the system and buy a pre-packaged system - RingCentral or some such.

There are no local asterisk professionals who can help, and we are a little
leery of opening up our system to outside consultants.

Anyone else face the above, and finally abandoned Asterisk for a commercial
system?

We have 167 users.
I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
conference rooms.

Suggestions welcome.

Best

Eddie
-- 
Eddie H. Mikell
Senior Systems Engineer
RKG

Office: 434.970.1010 x 124
Email: emikell at rimmkaufman.com

-- 
 <http://www.rimmkaufman.com>
<http://twitter.com/rimmkaufman> 
<http://www.linkedin.com/company/85385>
<http://plus.google.com/104980442218952272663/posts>
  <http://www.facebook.com/rimmkaufman>  <http://www.RKGblog.com>


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asterisk-users-bounces at lists.digium.com wrote on 10/28/2013 01:29:13 PM:
> From: Eddie Mikell <emikell at rimmkaufman.com>
> To: asterisk-users at lists.digium.com, 
> Date: 10/28/2013 01:29 PM
> Subject: [asterisk-users] Tired of dropouts and garbled phone calls 
> - where to go next?
> Sent by: asterisk-users-bounces at lists.digium.com
> 
> All,
> 
> The users in our organization are well, quite frankly, sick of phone
> service that is being provided.  The choppy phone calls, and drop 
> outs are detrimental to our sales force.
> 
> I've tried about everything I can think of.  
> 
> Moved the asterisk server from VM machine to dedicated machine
> More than enough bandwidth
> Setting 802.1p = 7
> Set Dedicated voice traffic 35% of bandwidth.
> 
> Not sure what option would be the best
> 
> Put analog lines in the conference room to avoid the dropouts - 
> leave the sip lines in place for day to day use
> Hire a consultant
> Ditch the system and buy a pre-packaged system - RingCentral or some 
such.
> 
> There are no local asterisk professionals who can help, and we are a
> little leery of opening up our system to outside consultants.
> 
> Anyone else face the above, and finally abandoned Asterisk for a 
> commercial system?  
> 
> We have 167 users.
> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
> conference rooms.
> 
> Suggestions welcome.
> 
> Best
> 
> Eddie
Does the garbled audio and dropouts only occur on outside calls, or do you 
get them on calls between extensions? How is your phone service delivered 
to your site?

If the extension to extension calls are clear, you need to be looking at 
your phone service and how you connect to it. If your local extensions are 
not clear, then you need to look at your asterisk implementation and your 
network.

If your incoming circuits are delivered via SIP, are you sure you have end 
to end QoS between your office and your phone provider? You can set all 
the QoS you want on the packets as they leave your network, but if your 
provider isn't honoring the QoS packets, then you can easily have audio 
issues. One of my locations has SIP service provided over a DSL line from 
the SIP provider. Just last week, the DSL line went down. We routed the 
calls out our standard internet connection and while it did work, we had 
audio dropouts (though only on incoming audio, the other end could hear us 
just fine). As soon as the DSL line was fixed and we routed back over 
their network, all the audio cleared up. It is the difference between low 
ping times and good QoS and higher ping times and providers who may not 
honor QoS.
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I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the 
Asterisk - No analogue.
We have a very lightly loaded 60 Mbs cable link to the Internet that 
tests pretty close to that most of the time.

I have not found any good tools to track down the causes of poor voice 
quality.
In my case, I have good incoming quality and terrible quality going out.
That is, I can hear people perfectly well but they complain that my 
voice drops out and is garbled regardless of who places the call.
As a result,  I use Skype for all of my calls and if someone calls me, I 
call them back on Skype if they have any problems.
I don't understand why Skype works so well and Asterisk works so poorly 
on the same environment.

Googling "Asterisk poor audio quality" return several hundred thousand
references

Ron
On 28/10/2013 2:29 PM, Eddie Mikell wrote:
> All,
>
> The users in our organization are well, quite frankly, sick of phone 
> service that is being provided.  The choppy phone calls, and drop outs 
> are detrimental to our sales force.
>
> I've tried about everything I can think of.
>
>     Moved the asterisk server from VM machine to dedicated machine
>
>     More than enough bandwidth
>
>     Setting 802.1p = 7
>
>     Set Dedicated voice traffic 35% of bandwidth.
>
> Not sure what option would be the best
>
>
>     Put analog lines in the conference room to avoid the dropouts -
>     leave the sip lines in place for day to day use
>
>     Hire a consultant
>
>     Ditch the system and buy a pre-packaged system - RingCentral or
>     some such.
>
> There are no local asterisk professionals who can help, and we are a 
> little leery of opening up our system to outside consultants.
>
> Anyone else face the above, and finally abandoned Asterisk for a 
> commercial system?
>
> We have 167 users.
> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the 
> conference rooms.
>
> Suggestions welcome.
>
> Best
>
> Eddie
> -- 
> Eddie H. Mikell
> Senior Systems Engineer
> RKG
>
> Office: 434.970.1010 x 124
> Email:emikell at rimmkaufman.com <mailto:emikell at rimmkaufman.com>
>
> <http://www.rimmkaufman.com>
> <http://twitter.com/rimmkaufman> 
> <http://www.linkedin.com/company/85385> 
> <http://plus.google.com/104980442218952272663/posts> 
> <http://www.facebook.com/rimmkaufman> <http://www.RKGblog.com>
>
>
>
>

-- 
Ron Wheeler
President
Artifact Software Inc
email: rwheeler at artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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On 10/28/2013 3:59 PM, Ron Wheeler said:
> I am reaching the same level of frustration.
> I have tried to find the source of the problems.
> We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk -
No analogue.
I don't have any problems with IAX, but I hear some do.
> We have a very lightly loaded 60 Mbs cable link to the Internet that tests
pretty close to that most of the time.
Bandwidth is less important than the overall quality of the internet link,
latency and jitter. Either way, there is no QoS on the internet, all bets are
off.

The codec can matter too. What are you using?
>
> I have not found any good tools to track down the causes of poor voice
quality.
> In my case, I have good incoming quality and terrible quality going out.
Oh, is your cable connection assymetric? Upload smaller than download? If so,
that correlates to terrible audio, right?
> That is, I can hear people perfectly well but they complain that my voice
drops out and is garbled regardless of who places the call.
> As a result,  I use Skype for all of my calls and if someone calls me, I
call them back on Skype if they have any problems.
> I don't understand why Skype works so well and Asterisk works so poorly
on the same environment.
>
> Googling "Asterisk poor audio quality" return several hundred
thousand references
I'd not shoot asterisk yet. I'd focus on the internet connection and
it's components (cable modem) first.

I use asterisk all over the place. Mostly connected to PRI's and Carrier
provided SIP trunks, with internet SIP trunks as backup. I get complaints on the
Internet based SIP trunks sometimes, never on other other two.

I'd ask most of these questions of the OP too. Overall telephony design
matters.



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On Mon, 28 Oct 2013, Eddie Mikell wrote:
> All,
> The users in our organization are well, quite frankly, sick of phone
service that is being provided. ?The choppy phone
> calls, and drop outs are detrimental to our sales force.
> 
> I've tried about everything I can think of. ?
>
>       Moved the asterisk server from VM machine to dedicated machine
>
>       More than enough bandwidth
>
>       Setting 802.1p = 7
>
>       Set Dedicated voice traffic 35% of bandwidth.
> 
> Not sure what option would be the best
>
>       Put analog lines in the conference room to avoid the dropouts - leave
the sip lines in place for day to day use
>
>       Hire a consultant
>
>       Ditch the system and buy a pre-packaged system - RingCentral or some
such.
> 
> There are no local asterisk professionals who can help, and we are a little
leery of opening up our system to outside
> consultants.
> 
> Anyone else face the above, and finally abandoned Asterisk for a commercial
system? ?
> 
> We have 167 users.
> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
conference rooms.
> 
> Suggestions welcome.
> 
> Best
> 
> Eddie
> --
As stated in previous replies if you haven't already I would certainly try 
to isolate the problem, e.g., are extension to extension calls good, is 
the problem only on outside calls etc.

We are starting our 4th year of VoIP service and have had two seemingly 
similar episodes to yours during that time.  We are on a non-symmetric 
cable connection, 20/4 (I believe).  After a few days of "crappy"
audio I
started looking for some way to characterize/correlate bad audio with 
something I could measure.  I found iperf (http://iperf.sourceforge.net/) 
to be a free and easy starting point, which actually turned out to be all 
I needed.

I simply ran a "server" instance on our "cloud" server
roughly 1K miles
away and a "client" instance locally.  I used the command line
swithces
that forced udp mode.  This allowed me to see jitter and packet loss in 
both directions.  We had terrible packet loss in the outbound direction. 
This didn't show up in normal browsing, emailing etc., kinds of things as 
I suspect TCP retries masked the problem.  With a little persistence with 
the cable company the second tech found a bad "tap" (I believe)
outside at
the cable drop.  Replacing that solved our issue for almost two years.

The next time this happened iperf showed a similar packet loss problem. 
This time it turned out to be "noise in the system" according to the
cable
tech.  He said it could be from any number of sources but a different team 
would be out to hunt it down the next day.  In the mean time he changed 
out our old Moto SB5101 modem for a more modern DOCSIS 3.0 modem.  The 
multiple channel bonding that it offered was much better at punching 
through the noise.  That change alone ended crappy audio as well as packet 
loss as shown by iperf.

On 10/28/2013 07:29 PM, Eddie Mikell wrote:
> All,
>
> The users in our organization are well, quite frankly, sick of phone
> service that is being provided.  The choppy phone calls, and drop outs
> are detrimental to our sales force.
>
> I've tried about everything I can think of.
>
>     Moved the asterisk server from VM machine to dedicated machine
That's a good start. Now what have you done to conclude that the 
Asterisk server is not the cause of your problems?
>     More than enough bandwidth
That's irrelevant. It's about the quality of that bandwidth. Have you 
figured out if there might be a lot of packetloss or are you perhaps on 
a cablelink which is a *shared* medium? Once your link hits the box in 
the street it shares it with others who might be eating up all the 
bandwidth with their torrent downloads etc.? Use tools like iperf, smoke 
ping and mtr to see if there are obvious problems on the route to your 
VoIP provider.
>     Setting 802.1p = 7
>
>     Set Dedicated voice traffic 35% of bandwidth.
>
> Not sure what option would be the best
Once the packets leave your premises and your ISP/cable company starts 
messing with them a QoS setting is generally not honored so not very 
helpful unless your LAN is congested.
>     Put analog lines in the conference room to avoid the dropouts -
>     leave the sip lines in place for day to day use
If those analog lines are cheap, easy to get then as an intermediate 
solution I would order those analog lines as fast as I could. Or fix the 
VoIP problems, whichever is faster.
>     Hire a consultant
An experienced VoIP consultant should be able to tell you what is or 
could be causing your problems. With your users "sick of phone
service"
it suprises me that you haven't already hired one.
>     Ditch the system and buy a pre-packaged system - RingCentral or some
>     such.
And what if it's your Internet link or the route to your VoIP provider? 
What if your VoIP provider is messing up?
> There are no local asterisk professionals who can help, and we are a
> little leery of opening up our system to outside consultants.
If you don't want that then you don't want that but given the state your
users are in I would be less worried about giving a Consultant access to 
the Asterisk box and more worried about my job :-)
> Anyone else face the above, and finally abandoned Asterisk for a
> commercial system?
I have seen that once years ago where some clueless sales guy had 
totally oversold an ancient Asterisk/Bristuff/ISDN setup which was very 
buggy and crash prone. There was no way to make that work reliably. 
After the supplier failed for months I was brought in to review the 
setup and possibly fix it. Told the customer to cut its losses. So they 
kicked out their supplier and opted for a different setup.
> We have 167 users.
> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
> conference rooms.
I don't know how Grandstream is these days. I thought the GXP2100 was ok 
but I guess you already know if there's a problem with those phones from 
the (lack of) intra-office call complaints from your users.
> Suggestions welcome.
Hire a Consultant or someone who has been part of this Community for a 
while and is well known on this list or in #asterisk on irc. Provide 
remote access if required. Change passwords afterwards.

If you really don't want to provide remote access then find a reputable 
VoIP provider with a switch physically as close as possible to your 
location, get a DID for a few bucks, hook it up to your Asterisk box and 
route it to a line on your phone, grab your cell, call that DID and see 
if you still have the problem. It wouldn't be the first time that the 
link between you and your VoIP provider just doesn't cut it. Or maybe 
your VoIP provider just sucks and you need to change to a different one. 
Both flowroute.com and voip.ms work well for me (no affiliation). Or 
maybe your Internet link sucks and you need to change your ISP.

Good luck.

Regards,
Patrick

On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote:
> All,
> 
> 
> The users in our organization are well, quite frankly, sick of phone
> service that is being provided.  The choppy phone calls, and drop outs
> are detrimental to our sales force.
> 
> 
> I've tried about everything I can think of.  
> 
> 
>         Moved the asterisk server from VM machine to dedicated machine
>         More than enough bandwidth
>         Setting 802.1p = 7
>         Set Dedicated voice traffic 35% of bandwidth.
>         
>         
> Not sure what option would be the best
>         
>         
>         Put analog lines in the conference room to avoid the dropouts
>         - leave the sip lines in place for day to day use
>         Hire a consultant
>         Ditch the system and buy a pre-packaged system - RingCentral
>         or some such.
>         
>         
> There are no local asterisk professionals who can help, and we are a
> little leery of opening up our system to outside consultants.
> 
> 
> Anyone else face the above, and finally abandoned Asterisk for a
> commercial system?  
> 
> 
> We have 167 users.
> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
> conference rooms.
> 
> 
> Suggestions welcome.
> 
> 
A general rule of thump after several years with voip

Voip turns out to be the "canary in the coal-mine" of a network. The
smallest change or problem will manifest itself as a voip issue no
matter what.


Now to some practical advice

Voip was designed for LAN's, The moment voip packets leave your lan and
go into a WAN of any sort, it could be the source of frustration for
many reasons.

1) Lots of routers/modems are not build to handle intense voip traffic.
voip generates lots of small in size UPD packages. In most of the cases
the routers/modems bridging your lan with the wan have no problem
handling them BUT what i have found is that once you get over a
threshold of traffic its possible the routers/modem can not cope with
it, mainly because the large number of packets they have to process.
In most enterprise grade routers the specs give you 2 numbers for the
size of data the router can handle.
total throughput and pps (packets per second). 
Usually total throughput is calculated using a packet size of around
1500bytes and it takes the router the same resources to process a 1500
bytes package as it does a 90bytes packet of a g729 call, as it just
looks at the headers and not the payload.So yes your router can handle
60Mbits (of 1500byte frames) which is about 5000 packers per second but
for voip that translates to less than 4Mbits of data (5000 packets of 90
bytes) 
I think you can get the picture


2) Because of 1) its possible that your ISP has issues, especially if
its handling lots of voip traffic while its equipment is not optimized
for that.

 
3) QOS and queing in general
Whatever you do with QOS to get a better priority/quality, the dirty
secret is, you can only control what YOU send, not what you receive.
And even that is true till your modem/router. Once the packet is gone
you have no control of how it will be handle by all intermediates till
it reaches its destination.
You have no idea if qos is honored by ALL hops and what kind of queuing
they apply (if they do) to that port/service/qos mark
That beeing said, its possible that you *might* have much better luck
with sip and sip rtp than with iax rtp  if your isp and all its
interconnects bother to offer qos for rtp.
Now for receiving it can be even harder if your isp does not provide
correct priority queuing for the rtp stream, as latencies can build fast
especially on "busy hours" (which happen to be the same hours people
use
their phones the most...) where people download stuff,emails etc.

ping.icmp and all the other networking monitoring tools/protocols could
be an indicator BUT its most probable that they will be handled by the
isp and its interconnects at the higher qos priority
The only way to see how rtp traffic is handled is to run rtp traffic.  

The only way around this is a "dedicated circut" MPLS or similar
between
the points of interest (i.e offices), with specific SLA which usually
means much much higher costs.
> 
> 
Finally my 2 cents for troubleshouting.
Check the network first !
Find what triggers the problem. 
Is it something that happens all time regardless of traffic ?
is it periodic ? (when bw goes over X percent, or at a specific time of
day ?)
Try different qos settings/priority queuing  on the
router
> 
-- 
Stelios S. Koroneos

Phone 
US : (+1) 347-783-5467
Greece : (+30) 211-800-7655 ext 101

Skype : skoroneos

PGP Key fingerprint = DC66 109A 6C3A 2D65 BA52  806E 6122 DAF4 32E7 076A

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