I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk eXchange driver definition ; [general] ; Specify bandwidth of low, medium, or high to control which codecs are used ; in general. ; bandwidth=low ; ; You can also fine tune codecs here using "allow" and "disallow" clauses ; with specific codecs. Use "all" to represent all formats. ; disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=gsm ; Always allow GSM, it's cool :) jitterbuffer=yes dropcount=3 maxjitterbuffer=500 maxexcessbuffer=100 minexcessbuffer=10 jittershrinkrate=1 register => XXXXXXXX:xxxxxxx@iaxtel.com ; Finally, you can set values for your TOS bits to help improve ; performance. Valid values are: ; lowdelay -- Minimize delay ; throughput -- Maximize throughput ; reliability -- Maximize reliability ; mincost -- Minimize cost ; none -- No flags ; tos=lowdelay but I still have a less-than-acceptible quality connection. The bandwidth usage is right around 5.5-6kbps. I have a multivoip that ran at about that rate and sounded fine (obviously with a different codec, but my point is that my broadband connection shouldn't be the issue). Any helpful suggestions? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot.
Had this problem earlier this week - ensure 'trunk=no' in iax.conf if
you're
using fairly current CVS code. There is something not right w/the trunking
that causes choppy sound. See the wiki for more info.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
-----Original Message-----
From: Michael George [mailto:george@mutualdata.com]
Sent: August 27, 2004 11:58 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] iaxtel and jitterbuffer
I am trying to work out IAX <--> IAX communications with my * box. I have
a
registration with iaxtel and I thought I would start there for my learning.
I am able to call the number for Digium's support line (700-428-6000), but
the
sound is horribly chopping. Some reading revealed the jitterbuffer
settings,
so I enabled them in iax.conf. I have the following now:
{clip}
Is timestamp information calculated purely from the relative timestamps of
each frame of the current incoming stream or is there some degree of RTC
synchronization expected between the two endpoints?
Similarly, are jitter calculations made seperately for each discrete channel
(ie. the IAX level) or are they based on an aggregate of all channels
between each pair of two endpoints (ie. the TCP/IP level)?
k.
-----Original Message-----
From: steve@daviesfam.org [mailto:steve@daviesfam.org]
Sent: August 29, 2004 12:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iaxtel and jitterbuffer
{clip}
The jitter buffer makes all its decisions about dejittering based on the
timestamps of incoming frames. There a fundamental expectation that the
sending side is correctly stamping each frame - 20msec, 40msec etc etc.
The problem is that the sending side doesn't always do that. Sometimes
for one reason or another the stamps "jump". The receiver has no way
of
telling that the sender mangled the timestamps, and assumes that the
packets with the new stamps have been delayed, or arrived early, or
whatever. Either way, the jitter buffer does its thing and unknowingly
makes things worse.
Unfortunately, this is why you can still be better off without it - but
the problem really needs to be fixed by fixing the timestamp generation on
the sender.
Steve
Just strip 00 and replace with 011
eg. Dial(SIP/your-US-provider/011${EXTEN:2}) etc
This will strip the leading 00 and replace with 011
Rgds
Tim
Johannes van Hulst wrote:
>I am looking for a way to convert numbers so that I can use one standard for
>all the providers.
>
>For example
>I would like to dail 00 31 20 1234567
>And that convert to 0011 31 20 1234567 so that my VS provider understands
>it.
>
>How can I remove the 00 and replace it with 0011.
>
>Greetings Han
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
Johannes van Hulst [Han.vanHulst@Terra.com.br] wrote:> I am looking for a way to convert numbers so that I can use one standard > for all the providers. > > For example > I would like to dail 00 31 20 1234567 > And that convert to 0011 31 20 1234567 so that my VS provider understands > it. > > How can I remove the 00 and replace it with 0011. >You could use "0011${EXTEN:2}", which would strip the first two digits ("00", in this case) and add "0011" to the start. -- _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/ _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h _/ _/ _/ _/ _/ _/ _/ _/_/ kevin@cursor.biz _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/
On Sun, 29 Aug 2004, Kris Boutilier wrote:> Is timestamp information calculated purely from the relative timestamps of > each frame of the current incoming stream or is there some degree of RTC > synchronization expected between the two endpoints?No sync is needed; its all relative.> Similarly, are jitter calculations made seperately for each discrete channel > (ie. the IAX level) or are they based on an aggregate of all channels > between each pair of two endpoints (ie. the TCP/IP level)?De-jtter is done for each call independently. Steve