From: Michael K. <mic...@ip...> - 2007-01-24 20:36:14
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I have an ASTLinux (development) system running at home of which I am quite happy with, however even the slightest download from one of the devices on the network causes voice degredation during a call due to packet loss. Frustrated with this, I set a phone directly to the ITSP and was unable to cause this packet loss issue, even with quite substantial downloads. I suspected that this was a jitter buffer issue in Asterisk but when I began looking at how to tune this, I was surprised to find that a dynamic jitter buffer was not actually available in SIP until 1.4. I understand that a dynamic jitter buffer has been available for IAX for a while. Is there any way I can solve this problem (install patch etc) or do I need to use 1.4. PS I would probably wait until it is included in ASTLinux, the best Asterisk distribution. Regards Michael Knill -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.8/649 - Release Date: 23/01/2007 8:40 PM |