Re: [asterisk-users] Audio dropping
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: 28 May, 2011 23:50 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Audio dropping On Fri, May 27, 2011 at 10:31:57AM +0200, Mark Scholten wrote: >What could the reason be audio in 1 direction is dropping? (Normally >from the Asterisk server to the mentioned SIP clients.) No clear >information is in the logs (it is like the call ended normally) and not >all calls are having problem (most not, but it happens to often for us >to start using VoIP more at the moment). While the most usual problem is packet filtering / NAT, this generally manifests as no audio at all in one direction, not a drop in mid-call. But it's possible that one of the intermediate transit providers is doing something "clever". (Disabling ping, as you mention in your later email, is often a good indicator of a company with insufficient Clue.) Are you in a position to tunnel the traffic over a VPN or similarly flat and unfilterable network link? (This might be a good idea anyway.) Hello Roger, I'm not in a position to start doing that. The ping is disabled on the firewall that is also doing the NAT (not something the provider does). However the provider is blocking other things (incoming email for example, for outgoing it is something I can understand). The strange thing is it is dropping mid-call, that is the reason I don't think it is only a NAT problem. Another network we use offered us a server so we can test if it is related to the other network we use. Regards, Mark -- _ -- 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
Re: [asterisk-users] Audio dropping
On Fri, May 27, 2011 at 10:31:57AM +0200, Mark Scholten wrote: >What could the reason be audio in 1 direction is dropping? (Normally from >the Asterisk server to the mentioned SIP clients.) No clear information is >in the logs (it is like the call ended normally) and not all calls are >having problem (most not, but it happens to often for us to start using VoIP >more at the moment). While the most usual problem is packet filtering / NAT, this generally manifests as no audio at all in one direction, not a drop in mid-call. But it's possible that one of the intermediate transit providers is doing something "clever". (Disabling ping, as you mention in your later email, is often a good indicator of a company with insufficient Clue.) Are you in a position to tunnel the traffic over a VPN or similarly flat and unfilterable network link? (This might be a good idea anyway.) Roger -- _ -- 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
Re: [asterisk-users] Audio dropping
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 27 May, 2011 10:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audio dropping On Fri, 2011-05-27 at 10:31 +0200, Mark Scholten wrote: > Hello, > > We see some strange behavior with phone calls, we use Asterisk 1.8.3.3. > > SIP clients (all behind NAT at different locations, so not a single > NAT solution is used): > - x-lite > - linksys pap2t > - polycom kirk (multiple type numbers) > - polycom (multiple type numbers, hardware phones) > > Our Asterisk servers stays in between (some calls are recorded). > Asterisk is running on a physical server (no virtual server software) with "old" > hardware (Xeon 3.2 GHz with hypertrading and 4GB RAM, mainly used for > buffers). We use a MySQL backend (CDR records are stored in it and SIP > users are stored in a MySQL database). > > We use a SIP provider with a trunk for outgoing and incoming calls, > this is also an Asterisk server if I'm correct. We currently do around > 1000 calls a week and max. do 10 calls at the same time. The Asterisk > server is not behind a NAT. > > What could the reason be audio in 1 direction is dropping? (Normally > from the Asterisk server to the mentioned SIP clients.) No clear > information is in the logs (it is like the call ended normally) and > not all calls are having problem (most not, but it happens to often > for us to start using VoIP more at the moment). > > To test if it was the firewall we disabled the firewall on the > Asterisk server and moved the Asterisk server before the other firewalls we have. > > What could the problem be? And even more important what could solve it > (and/or explain it)? > > Kind regards, > > Mark Scholten > Hi Are the broadband connections to the target SIP extensions dedicated for VoIP or does any other traffic run over them? We tend to find that 80% of call quality issues are caused by the broadband connection. A good diagnostic tool to keep an eye on the broadband connections involved is smokeping http://oss.oetiker.ch/smokeping/ We find it an absolute godsend. === Hello, Ping is disabled on the location with most call problems (filtered). A few clients with dedicated audio broadband connections have the same problem compared to "shared" broadband connections. This makes it very difficult to find the problem. A supplier we use offered to test it with a server from them, if that solves it it was probably a network issue. Regards, Mark -- _ -- 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
Re: [asterisk-users] Audio dropping
On Fri, 2011-05-27 at 10:31 +0200, Mark Scholten wrote: > Hello, > > We see some strange behavior with phone calls, we use Asterisk 1.8.3.3. > > SIP clients (all behind NAT at different locations, so not a single NAT > solution is used): > - x-lite > - linksys pap2t > - polycom kirk (multiple type numbers) > - polycom (multiple type numbers, hardware phones) > > Our Asterisk servers stays in between (some calls are recorded). Asterisk is > running on a physical server (no virtual server software) with "old" > hardware (Xeon 3.2 GHz with hypertrading and 4GB RAM, mainly used for > buffers). We use a MySQL backend (CDR records are stored in it and SIP users > are stored in a MySQL database). > > We use a SIP provider with a trunk for outgoing and incoming calls, this is > also an Asterisk server if I'm correct. We currently do around 1000 calls a > week and max. do 10 calls at the same time. The Asterisk server is not > behind a NAT. > > What could the reason be audio in 1 direction is dropping? (Normally from > the Asterisk server to the mentioned SIP clients.) No clear information is > in the logs (it is like the call ended normally) and not all calls are > having problem (most not, but it happens to often for us to start using VoIP > more at the moment). > > To test if it was the firewall we disabled the firewall on the Asterisk > server and moved the Asterisk server before the other firewalls we have. > > What could the problem be? And even more important what could solve it > (and/or explain it)? > > Kind regards, > > Mark Scholten > Hi Are the broadband connections to the target SIP extensions dedicated for VoIP or does any other traffic run over them? We tend to find that 80% of call quality issues are caused by the broadband connection. A good diagnostic tool to keep an eye on the broadband connections involved is smokeping http://oss.oetiker.ch/smokeping/ We find it an absolute godsend. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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
[asterisk-users] Audio dropping
Hello, We see some strange behavior with phone calls, we use Asterisk 1.8.3.3. SIP clients (all behind NAT at different locations, so not a single NAT solution is used): - x-lite - linksys pap2t - polycom kirk (multiple type numbers) - polycom (multiple type numbers, hardware phones) Our Asterisk servers stays in between (some calls are recorded). Asterisk is running on a physical server (no virtual server software) with "old" hardware (Xeon 3.2 GHz with hypertrading and 4GB RAM, mainly used for buffers). We use a MySQL backend (CDR records are stored in it and SIP users are stored in a MySQL database). We use a SIP provider with a trunk for outgoing and incoming calls, this is also an Asterisk server if I'm correct. We currently do around 1000 calls a week and max. do 10 calls at the same time. The Asterisk server is not behind a NAT. What could the reason be audio in 1 direction is dropping? (Normally from the Asterisk server to the mentioned SIP clients.) No clear information is in the logs (it is like the call ended normally) and not all calls are having problem (most not, but it happens to often for us to start using VoIP more at the moment). To test if it was the firewall we disabled the firewall on the Asterisk server and moved the Asterisk server before the other firewalls we have. What could the problem be? And even more important what could solve it (and/or explain it)? Kind regards, Mark Scholten -- _ -- 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
