[asterisk-users] Issue call quality: Asterisk call quality on trunks

[Henry Fernandes]

It¹s not clear to me if you¹ve done troubleshooting to determine where the
quality issues are occurring.  Try testing outbound/external calls
separately from internal calls (i.e., calls that stay on your network and
don¹t go out over the trunk to the carrier).

If the problem is on internal calls, then I¹d say the quality issues are
caused by something local ‹ perhaps a networking issue or an issue with
virtualization.  If the problem is on internal calls, you should also test
calls without any codec conversion.

If the problem is on outbound/external calls only, then the issue might be
something on the LAN or ISP side.  As a first step, you could get a VoIP
Spear (voipspear.com) account and see if you can notice any problems with
that.  A second step would be to get packet captures.
-H

From:  Kristof Van Den Ouweland <kvandenouweland at vangenechten.com>
Reply-To:  Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Date:  Monday, July 6, 2015 at 11:58 PM
To:  <asterisk-users at lists.digium.com>
Subject:  [asterisk-users] Issue call quality: Asterisk call quality on
trunks

Good afteroon all,

First of all: thanks for everybody who is willing to think this through with
me:

I'm having some issues regarding call quality between some calls. Let me try
to explain the situation first

We have a Asterisk 11.16 server based on the Xivo distribution. There are 2
servers running in cluster (Active Passive), both virtual with the following
config:
Quadcore CPU
8 GB ram
About 50Gb of diskspace which is used for about 15%
(Let's call this Asterisk cluster 001 for clarity)

The Asterisk server has a trunk to a cisco call manager which is on the same
site/LAN, and 4 trunks to other Asterisk servers (same distribution but
lower specs, name Asterisk cluster 002 and 003). These are all sites in our
WAN but they are geographically divided and connected via MPLS links.  Each
affiliate has a specific number range XXXYYY where XXX stands for the
affiliate and YYY is the extension of the users.
(Average bandwidth = 4Mpbs which has to be shared by applications. QoS
allows that VoIP is prioritized)

Now, the actual problem:

I've set my main codecs to G711 a-law, G7 222 (for cisco call manager) and
GSM as last. The GSM is set as primary for those trunks which don't have 4
Mbps of bandwidth available.

In most cases, trunk calling results in bad quality of conversations (a-law
is chosen as codec)  but or it is jitterish, or one party does not hear the
other party (complete silence) It could be that the second time they call,
everything is ok.

--

So a little ASCII map about the geographical setup:

Aff 1: [Asterisk cluster 001] <-- LAN trunk --> Cisco call manager
                    |
                MPLS connection 20Mbps
                    |
                    |------>  MPLS Cloud    <---> MPLS connection 2Mbps -->
[Asterisk cluster 002]
                                                       | <---->  MPLS
connection 4 Mbps --> [Asterisk cluster 003]

Calls between Cluster 001 <---> cluster 002 or 003 are potentially of bad
quality (sometimes ok but most of all jiterish)
Calls between Cluster 002 <---> cluster 003 are good

The bandwidth if cluster001 ( 20 Mbps) is used about 50% with peaks to 75%.

I've aslo actived the jitter buffer with a buffer of 200ms but this didn't
seem to do any good.

Does anybody have some hints how I can troubleshoot this?

Note: the Cisco calls to the other affiliaters over the same WAN don't have
issues but these are based on SCCP protocol.

Thanks in advance
Kristof




 
 
 Privileged Confidential Information may be contained in this message. If
you are not the addressee indicated in this message (or responsible for
delivery of the message to such person), you may not copy or deliver  this
message to anyone.
In such case, you should destroy this message  and kindly notify the sender
by reply email.
Please advise  immediately if you or your employer does not consent to
Internet email  for messages of this kind.
Opinions, conclusions and other  information in this message that do not
relate to the official business  of my firm shall be understood as neither
given nor endorsed by it.
 
-- 
This message has been scanned for viruses and dangerous content by
Cisa Antispam Service, and is believed to be clean.
-- _____________________________________________________________________ --
Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or
update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150707/1fc3f27f/attachment.html>