Well, what have you done in your network switch? Is QoS enabled with DSCP 46 (ef) piped to the low latency queue?
From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tran, Ly V. Sent: Thursday, June 17, 2010 3:32 PM To: Josh Patten; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] QOS Best Practices for VOIP? Both the phone system and computers are all connected to a Cisco 3550 / 3560 switch. From there, to the Firebox. As far as QOS goes on the switches, I've been told that the default Auto QOS config is applied. I'm more of a server guy, so I'm not an expert at Cisco hardware.. I just need to find out what is the best config is and pass it along to our network person to configure. Our ITSP is Voxitas, but we are not using their MPLS service or have a dedicated connection to them. The Internet connection we have is a fiber DIA in which the ISP says every packet passes through, no QOS on their end and that we implement the tagging and it should pass through. QOS on the firebox is set with DSCP marking with an Assigned 46(EF) value on all the interfaces. I suppose we will have to match that on the switches. Ly Tran From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten Sent: Thursday, June 17, 2010 12:37 PM To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] QOS Best Practices for VOIP? There's nothing on the other side of the equals sign. You haven't told us anything about how QoS tagging is done internally before it reaches the firebox. Who is your ITSP? Are you tagging packets at the switch level for the voice VLAN? More details are necessary. On 06/17/2010 12:30 PM, Tran, Ly V. wrote: What are the essentials that one can do to improve call quality in using SipX? We have successfully replaced our Covad VOIP system with SipX. Covad was using MGCP and we had a 1.5M sDSL line dedicated to voice. In moving to SipX we installed a 10M DIA connection that comes to us through fiber, and are sharing both voice/data. However, we are still hearing from our users both remotely and internally that the voice quality can be poor sometimes.. breaking audio or jitter. Our setup consists of SipX using a siptrunk, behind a Watchguard Firebox. The phone system is on its own VLAN. I understand that once it enters the Internet, its beyond our control. We have configured the Watchguard for QOS, but not sure what the best configuration is for SipX... ie. DSCP? The phones we have are Cisco 7960s and LG 6812s.. are there specific settings we can do on these phones to improve quality? Ly Tran _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
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