On Thu, 12 Mar 2015 10:04:08 -0400, Andres wrote: > On 3/12/15 9:39 AM, Ron Wheeler wrote: >> Your characterization may be true but Skype works much better than SIP >> when it comes to sound quality. >> > SIP is not to blame for this. Its the audio codec being used. Skype has > spend a great deal of effort with their SILK codec by making it highly > tolerant of packet loss and jitter. The same cannot be said for the > standard codecs Asterisk uses. >> I have SIP softphone with Asterisk server and Skype on the same >> workstation. >> Skype just works better over the same network. >>
The thing to remember about Skype is that they started out as the small guy, and they had some very interesting ideas, IMHO. I don't actually know it's a sound quality issue, per say. It's double+ NAT, with a wi-fi bridge, plus, sometimes, another wi-fi network. In that situation, skype works from a cell phone! Granted, there are dropped calls, but, eh. The way things stand, I can't, unfortunately, use Ekiga to connect to the **outside** SIP provider because, apparently, there are too many hops: http://superuser.com/questions/880705/ IAX might be useful in this circumstance :) -Thufir -- _____________________________________________________________________ -- 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
