Re: [Asterisk-Users] LiveVOIP troubleshooting
I think it's a server/connection issue with the LiveVoip server. I'm connected to their Winnipeg server and I get pretty much perfect calling, all the time. A buddy of mine recently got setup on the Vancouver server and is also experiencing choppy audio. He's in the process of asking if he can get moved to the Winnipeg server. We'll see what happens Paul - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 02, 2005 7:21 PM Subject: [Asterisk-Users] LiveVOIP troubleshooting Hi everyone, I need some ideas to troubleshoot this issue: I recently got an 800 numbers from LiveVOIP and it works but on most calls the caller gets hears choppy audio (one drop out per 10 seconds or so). I know this isn't LiveVOIP's support forum but I'm sure some here use their 800 service and I'm interested in their feedback and ideas. And don't get me wrong, LiveVOIP's support has been quite good -- cooperative, fast response, action taken as requested -- but I do not want to try their patience. At this point I am not blaming them for this issue either. Here's the summary: * I'm connected via IAX2 to * The server is in a datacenter with plenty of bandwidth. * Using ulaw with standard 20 ms frames. * I hear the caller perfectly fine, caller hears choppy audio. * tcpdump shows incoming and outgoing packets right on time, every 20 ms in each direction. * I'm not using trunking for now. * Pings to LiveVOIP are about 35 ms. * iax2 show channels shows 1 ms jitter, 42 ms lag. * Drop outs occur on IVR (or audio generated on the server itself) or during normal conversation with a SIP client (ATA or phone) connected to the server remotely. Connection between server and phones is well tested and working fine. I have asked LiveVOIP to switch me from their Vancouver node to their New York node, which reduced ping times from 50 ms to 35 ms. Less chops but still not perfect. Note that the same server is already connected to several Broadvoice accounts, which work flawlessly. Anyway, if anyone has some ideas of what I can try, please let me know. I do not want to keep trying all their nodes to find one that works for me. I do not necessarily want to use a different codec either since I have the bandwidth and I may be receiving faxes, so I need ulaw. Thanks and sorry for the long-ish post. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVOIP troubleshooting
I think it's a server/connection issue with the LiveVoip server. Perhaps. I'm still testing, but since moving to their NYC server, audio quality has been very good to excellent in the past days. Not too bad, after all, but still too early to draw a conclusion. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LiveVOIP troubleshooting
Hi everyone, I need some ideas to troubleshoot this issue: I recently got an 800 numbers from LiveVOIP and it works but on most calls the caller gets hears choppy audio (one drop out per 10 seconds or so). I know this isn't LiveVOIP's support forum but I'm sure some here use their 800 service and I'm interested in their feedback and ideas. And don't get me wrong, LiveVOIP's support has been quite good -- cooperative, fast response, action taken as requested -- but I do not want to try their patience. At this point I am not blaming them for this issue either. Here's the summary: * I'm connected via IAX2 to * The server is in a datacenter with plenty of bandwidth. * Using ulaw with standard 20 ms frames. * I hear the caller perfectly fine, caller hears choppy audio. * tcpdump shows incoming and outgoing packets right on time, every 20 ms in each direction. * I'm not using trunking for now. * Pings to LiveVOIP are about 35 ms. * iax2 show channels shows 1 ms jitter, 42 ms lag. * Drop outs occur on IVR (or audio generated on the server itself) or during normal conversation with a SIP client (ATA or phone) connected to the server remotely. Connection between server and phones is well tested and working fine. I have asked LiveVOIP to switch me from their Vancouver node to their New York node, which reduced ping times from 50 ms to 35 ms. Less chops but still not perfect. Note that the same server is already connected to several Broadvoice accounts, which work flawlessly. Anyway, if anyone has some ideas of what I can try, please let me know. I do not want to keep trying all their nodes to find one that works for me. I do not necessarily want to use a different codec either since I have the bandwidth and I may be receiving faxes, so I need ulaw. Thanks and sorry for the long-ish post. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVOIP troubleshooting
switch to real provider On Mon, 2005-05-02 at 20:21, Luki wrote: Hi everyone, I need some ideas to troubleshoot this issue: I recently got an 800 numbers from LiveVOIP and it works but on most calls the caller gets hears choppy audio (one drop out per 10 seconds or so). I know this isn't LiveVOIP's support forum but I'm sure some here use their 800 service and I'm interested in their feedback and ideas. And don't get me wrong, LiveVOIP's support has been quite good -- cooperative, fast response, action taken as requested -- but I do not want to try their patience. At this point I am not blaming them for this issue either. Here's the summary: * I'm connected via IAX2 to * The server is in a datacenter with plenty of bandwidth. * Using ulaw with standard 20 ms frames. * I hear the caller perfectly fine, caller hears choppy audio. * tcpdump shows incoming and outgoing packets right on time, every 20 ms in each direction. * I'm not using trunking for now. * Pings to LiveVOIP are about 35 ms. * iax2 show channels shows 1 ms jitter, 42 ms lag. * Drop outs occur on IVR (or audio generated on the server itself) or during normal conversation with a SIP client (ATA or phone) connected to the server remotely. Connection between server and phones is well tested and working fine. I have asked LiveVOIP to switch me from their Vancouver node to their New York node, which reduced ping times from 50 ms to 35 ms. Less chops but still not perfect. Note that the same server is already connected to several Broadvoice accounts, which work flawlessly. Anyway, if anyone has some ideas of what I can try, please let me know. I do not want to keep trying all their nodes to find one that works for me. I do not necessarily want to use a different codec either since I have the bandwidth and I may be receiving faxes, so I need ulaw. Thanks and sorry for the long-ish post. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
