<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-family: Calibri, sans-serif;"><div>It’s not clear to me if you’ve 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 don’t go out over the trunk to the carrier).</div><div><br></div><div>If the problem is on internal calls, then I’d 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.</div><div><br></div><div>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.</div><div>-H</div><div><br></div><span id="OLK_SRC_BODY_SECTION"><div style="font-family:Calibri; font-size:11pt; text-align:left; color:black; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; BORDER-RIGHT: medium none; PADDING-TOP: 3pt"><span style="font-weight:bold">From: </span> Kristof Van Den Ouweland <<a href="mailto:kvandenouweland@vangenechten.com">kvandenouweland@vangenechten.com</a>><br><span style="font-weight:bold">Reply-To: </span> Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br><span style="font-weight:bold">Date: </span> Monday, July 6, 2015 at 11:58 PM<br><span style="font-weight:bold">To: </span> <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br><span style="font-weight:bold">Subject: </span> [asterisk-users] Issue call quality: Asterisk call quality on trunks<br></div><div><br></div><div><meta name="Generator" content="Novell Groupwise Client (Version 14.0.1 Build: 118418)"><meta http-equiv="Content-Type" content="text/html; charset=utf-8"><div style="font: 10pt/normal Segoe UI; margin: 16px 16px 4px; font-size-adjust: none; font-stretch: normal;"><div class="GroupWiseMessageBody" id="GroupWiseSection_1436187047000_kvandenouweland@vangenechten.com_26CE398115A00000AD5CCF5D93F29635_"><div>Good afteroon all,</div><div><br></div><div>First of all: thanks for everybody who is willing to think this through with me:</div><div><br></div><div>I'm having some issues regarding call quality between some calls. Let me try to explain the situation first</div><div><br></div><div>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:</div><div>Quadcore CPU</div><div>8 GB ram</div><div>About 50Gb of diskspace which is used for about 15%</div><div>(Let's call this Asterisk cluster 001 for clarity)</div><div><br></div><div>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.</div><div>(Average bandwidth = 4Mpbs which has to be shared by applications. QoS allows that VoIP is prioritized)</div><div><br></div><div>Now, the actual problem:</div><div><br></div><div>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.</div><div><br></div><div>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.</div><div><br></div><div>--</div><div><br></div><div>So a little ASCII map about the geographical setup:</div><div><br></div><div>Aff 1: [Asterisk cluster 001] <-- LAN trunk --> Cisco call manager</div><div> | </div><div> MPLS connection 20Mbps</div><div> |</div><div> |------> MPLS Cloud <---> MPLS connection 2Mbps --> [Asterisk cluster 002]</div><div> | <----> MPLS connection 4 Mbps --> [Asterisk cluster 003] </div><div><br></div><div>Calls between Cluster 001 <---> cluster 002 or 003 are potentially of bad quality (sometimes ok but most of all jiterish)</div><div>Calls between Cluster 002 <---> cluster 003 are good </div><div><br></div><div>The bandwidth if cluster001 ( 20 Mbps) is used about 50% with peaks to 75%.</div><div><br></div><div>I've aslo actived the jitter buffer with a buffer of 200ms but this didn't seem to do any good.</div><div><br></div><div>Does anybody have some hints how I can troubleshoot this?</div><div><br></div><div>Note: the Cisco calls to the other affiliaters over the same WAN don't have issues but these are based on SCCP protocol.</div><div><br></div><div>Thanks in advance</div><div>Kristof</div><div><br></div><div><br></div><div><br></div></div><br>
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