Re: [Asterisk-Users] Sound quality - commercial vs. Asterisk
Paul Fielding wrote: So far in my playing with Asterisk I've messed with soft phones (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters (Grandstream 286, Digium IAXy). I've also got a Vonage line, using a Linksys ATA. None of the devices I've connected to my Asterisk server have been able to maintain the same consistent sound quality over a long distance as the Vonage line.Don't get me wrong, the Grandstreams are actually not too bad, but there is still some breakups that can be annoying. Meanwhile the Vonage ATA maintains an almost flawless connection, all the time. I'm assuming (perhaps wrongly?) that the Linksys ATA that Vonage uses is still using SIP with some standardized codec. If that assumption is correct, then how the heck to they manage to get the consistent connection quality? Is it just a matter of the right setting tweaks within Asterisk and/or the SIP devices? I don't think it's a question of Asterisk hardware, since if I connect via local network to the Asterisk server with a SIP device the quality is pretty consistent. It's generally when remotely connecting that I have the inconsistent sound quality. This would lead me to believe that it's a matter of tweaking something to deal with latency or packet dropping issues (?). What has Vonage got figured out that I still need to? Any comments would be appreciated... regards, Paul Likely, you are running into packet queue problems. As I recall, the vonage device goes on the line before anything else, so it can shape the stream to put it's bits first, ensuring it's packets get out in a timely matter ( #1 important thing in voip ). If you were to shape your stream and put your voip bits first, then I think you'd see an improvement in the qualty of service. Granted, I don't know your particular situation, so this could all be guess work. Sean ___ 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] Sound quality - commercial vs. Asterisk
Paul Fielding wrote: So far in my playing with Asterisk I've messed with soft phones (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters (Grandstream 286, Digium IAXy). I've also got a Vonage line, using a Linksys ATA. None of the devices I've connected to my Asterisk server have been able to maintain the same consistent sound quality over a long distance as the Vonage line. Don't get me wrong, the Grandstreams are actually not too bad, but there is still some breakups that can be annoying. Meanwhile the Vonage ATA maintains an almost flawless connection, all the time. I'm assuming (perhaps wrongly?) that the Linksys ATA that Vonage uses is still using SIP with some standardized codec. If that assumption is correct, then how the heck to they manage to get the consistent connection quality? Is it just a matter of the right setting tweaks within Asterisk and/or the SIP devices? I don't think it's a question of Asterisk hardware, since if I connect via local network to the Asterisk server with a SIP device the quality is pretty consistent. It's generally when remotely connecting that I have the inconsistent sound quality. This would lead me to believe that it's a matter of tweaking something to deal with latency or packet dropping issues (?). A better jitterbuffer and Packet Loss Concealment is what you need. It's coming to asterisk soon. http://bugs.digium.com/bug_view_page.php?bug_id=0002532 ___ 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] Sound quality - commercial vs. Asterisk
- Original Message - From: Sean Kennedy [EMAIL PROTECTED] Likely, you are running into packet queue problems. As I recall, the vonage device goes on the line before anything else, so it can shape the stream to put it's bits first, ensuring it's packets get out in a timely matter ( #1 important thing in voip ). If you were to shape your stream and put your voip bits first, then I think you'd see an improvement in the qualty of service. I agree I probably am having some packet queue problems, however i don't think it's my only problem. My Vonage ATA adapter is actually further behind the line than my Asterisk server. My configuration is such: Cable Modem -- Asterisk Server Linux Router (Each have their own real IP) -- Vonage ATA (behind Linux router) I don't have QoS running on my Linux box, though I've been thinking about trying to implement it. If I did manage to get it implemented then I'll probably also move my Asterisk server behind the router. Up until now I've left the Asterisk server in the real world due to problems running Asterisk behind a NAT. Those problems, however, seem to have been dealt with and a friend of mine is successfully running his behind a NAT.So QoS may be the way to, perhaps. I don't think it'll resolve all my issues though - if the Vonage ATA can do it withough QoS running, then surely there's more I can do with Asterisk. Perhaps the new jitter buffering coming soon will fix... *shrug*... Paul ___ 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] Sound quality - commercial vs. Asterisk
Paul Fielding wrote: I agree I probably am having some packet queue problems, however i don't think it's my only problem. My Vonage ATA adapter is actually further behind the line than my Asterisk server. My configuration is such: I wonder about codec-related issues, as well. Any chance that the Cisco adapter is using g711 and the other phone is using something with a lesser sound quality, or that perhaps some transcoding is going on that might be introducing some quality degradation? B. ___ 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] Sound quality - commercial vs. Asterisk
- Original Message - From: Brian Capouch [EMAIL PROTECTED] Paul Fielding wrote: I agree I probably am having some packet queue problems, however i don't think it's my only problem. My Vonage ATA adapter is actually further behind the line than my Asterisk server. My configuration is such: I wonder about codec-related issues, as well. Any chance that the Cisco adapter is using g711 and the other phone is using something with a lesser sound quality, or that perhaps some transcoding is going on that might be introducing some quality degradation? I'm not sure what codec the Vonage ATA (Linksys) is using, though I'd be interested to find out. When using my Asterisk server, either using Grandstream BT-101/2 or X-Ten Pro, I'm using g711. In some cases it's g711 to g711, in other's it's g711 to a ZAP chanel to an outside line. I don't think there's any other transcoding going on Paul ___ 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] Sound quality - commercial vs. Asterisk
So far in my playing with Asterisk I've messed with soft phones (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters (Grandstream 286, Digium IAXy). I've also got a Vonage line, using a Linksys ATA. None of the devices I've connected to my Asterisk server have been able to maintain the same consistent sound quality over a long distance as the Vonage line. Don't get me wrong, the Grandstreams are actually not too bad, but there is still some breakups that can be annoying. Meanwhile the Vonage ATA maintains an almost flawless connection, all the time. I'm assuming (perhaps wrongly?) that the Linksys ATA that Vonage uses is still using SIP with some standardized codec. If that assumption is correct, then how the heck to they manage to get the consistent connection quality? Is it just a matter of the right setting tweaks within Asterisk and/or the SIP devices? I don't think it's a question of Asterisk hardware, since if I connect via local network to the Asterisk server with a SIP device the quality is pretty consistent. It's generally when remotely connecting that I have the inconsistent sound quality. This would lead me to believe that it's a matter of tweaking something to deal with latency or packet dropping issues (?). What has Vonage got figured out that I still need to? Any comments would be appreciated... regards, Paul ___ 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
