[asterisk-users] Call quality issue across VPN- POTS vs SIP
Hi All, Got a strange (at least IMHO) issue that doesn't make much sense to me. Basic configuration is two sites with a site-to-site (aka router-to-router) VPN. Both sites have Cisco 7961G phones [with SIP firmware] on users' desks, and the only VoIP is internal - all of our outward telecom is on POTS or Centrex-enabled POTS lines. Site 1 has a Dell PowerEdge 1950 with Asterisk built from source and an AEX804E to connect to the outside world. Site 2 has an Asterisk Appliance with the 4 FXO / 4 FXS configuration, with the FXS ports currently unused. The PBXes at each site are configured to be essentially independent but with a unified dial plan so that calls can be placed or transferred across the VPN with a SIP trunk connecting the two PBXes, and canreinvite=no is set everywhere. The only other heavy consumer of bandwidth across the VPN is a real-time file replication suite that we use for file synchronization. While this is the ultimate issue, I don't understand the phenomena I'm seeing: If a user dials in to one of Site 2 FXO lines then dials across the VPN to a user at Site 1 while the file replication job is running audio quality (to the caller on the POTS line only) is abysmal-- audibly it sounds like about 50% of the packets are dropped (He__ Th__ __u __r, T__s is ___ln) On the other hand if a user at Site 2 picks up one of the Cisco phones [with the replication job still running] and dials across the VPN to a user at Site 1 audio quality is fantastic, ditto if a user at Site 1 calls to a user at Site 2. Any ideas why the audio quality would be so markedly different when the only thing that seems to be different is where the call is originating from (POTS line vs. SIP phone)? Replacing the border gear with equipment that's QOS aware and can handle prioritization is already on the list (and may be in the process of being ordered at this point) Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP
On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: Any ideas why the audio quality would be so markedly different when the only thing that seems to be different is where the call is originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP
Bob, It's conceivable, but how would I verify this and how would I change it if that was the problem? The site that those calls terminate at is using an Asterisk Appliance so most of the config is done for you but it is possible to tweak the underlying configuration files (and I also have SSH access so I can do asterisk -v -r) -- If I know what I need to tweak. Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: Any ideas why the audio quality would be so markedly different when the only thing that seems to be different is where the call is originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP
Just as an interesting follow-up/additional information, if I place a call to Site 2 on a POTS line, someone at Site 2 answers the call (using one of the Cisco phones) and then transfers it to me across the VPN the call sounds fine. So I think Bob's question was on the right track with it being a CODEC issue, but I'm not sure how I need to deal with that for the ZAP channel type. Thanks again, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lincoln King-Cliby Sent: Monday, November 03, 2008 1:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP Bob, It's conceivable, but how would I verify this and how would I change it if that was the problem? The site that those calls terminate at is using an Asterisk Appliance so most of the config is done for you but it is possible to tweak the underlying configuration files (and I also have SSH access so I can do asterisk -v -r) -- If I know what I need to tweak. Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: Any ideas why the audio quality would be so markedly different when the only thing that seems to be different is where the call is originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality issue across VPN- POTS vs SIP
On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote: It's conceivable, but how would I verify this and how would I change it if that was the problem? There's a few things you can do here. 1) Check the sip.conf on both sides to see what is defined there for the trunk. Look for some disallow and allow statements. If they are there, that will tell Asterisk what codecs to use on that trunk. 2) You could also check the codec that is in use during a call by looking at the sip channel. From the asterisk CLI, start with show channel SIP/ and tab it out to complete the command showing the trunk between your two systems. I believe the codecs are listed here as NativeFormats and ReadFormat. You could check this under both of your scenarios to see if there is a different codec in use. 3) If you'd like to try and force the use of a compressed codec such as GSM between your two sites, you would just need to make sure that both sides had the following lines in the definition for the trunk in sip.conf and then do a 'reload chan_sip.so from the Asterisk CLI: disallow=all allow=gsm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
