Ok.. wow .03ms (if that's the default) is DEFINATELY part of the issue
(audio going from the phone over wireless is slightly choppy).. while
audio coming in (20ms) is ok... where do you change it on the sipura?<br><br><div><span class="gmail_quote">On Apr 3, 2005 4:07 PM, <b class="gmail_sendername">Bruce Komito</b> <<a href="mailto:brucek@bagel.com">brucek@bagel.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">The packet size is a function of the number of milliseconds of sound sent<br>in the RTP packet. I don't know how to force * to change this, but you<br>*can* unilaterally change the RTP packet size on the Sipura. By doing<br>this, RTP packets sent by the Sipura will be larger or smaller than the<br>default (.03 ms is the default), and I know * will swallow whatever the<br>Sipura sends it. So, I know it's possible to change this in at least one<br>direction if you are using a Sipura.<br><br>Bruce Komito<br>High Sierra Networks, Inc.<br><a href="http://www.servers-r-us.com">www.servers-r-us.com</a><br>(775) 236-5815<br><br><br>On Sun, 3 Apr 2005, Matt wrote:<br><br>> IAX is not an option as Sipura devices do not support AIX.<br>> Yes, the sipura will handle the different packet sizes...<br>><br>> Is it possible to reprogram asteris to do this?<br>><br>> On Apr 3, 2005 1:55 AM, Steven Critchfield <<a href="mailto:critch@basesys.com">critch@basesys.com</a>> wrote:<br>> ><br>> > On Sat, 2005-04-02 at 21:16 -0500, Matt wrote:<br>> > > I'm aware that asterisk only supports 20ms packetization rates. Due<br>> > > to the fact that I will be using some voip devices on a wireless<br>> > > network which is highly sensative to framerate.. is there any way I<br>> > > can hard code the packetization rate at say 30 or 40ms and then<br>> > > compile astrisk? If so, can anyone in the know tell me what variables<br>> > > I need to look at to change?<br>> ><br>> > Are you sure your other devices support different packet sizes? Are you<br>> > sure the added delay in audio delivery can be handled decently and not<br>> > cause added echo?<br>> ><br>> > Have you considered what IAX trunking can do for you? It will reduce<br>> > frame rate as you add channels since each packet will then hold the<br>> > frames for each of the consecutive calls.<br>> > --<br>> > Steven Critchfield <<a href="mailto:critch@basesys.com">critch@basesys.com</a>><br>> ><br>> ><br>><br>><br>> This message has been categorized as "Indeterminate" by Bayesian Analyzer.<br>> Please click on this link if this message is a Spam<br>> <a href="http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=2">http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=2</a><br>><br>> Or on this link if this message is a legitimate mail<br>> <a href="http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=1">http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-04-03%5C140ee012d55b40a08232629f70c89189&C=1</a><br>><br>><br>> --<br>> -----------------------------------------------------------------------<br>> This message has been inspected by DynaComm i:mail<br>> -----------------------------------------------------------------------<br>><br><br></blockquote></div><br>