Re: [Asterisk-Users] Voice over Frame Relay Asterisk

2005-03-05 Thread Rich Adamson
 Has anyone done Voice Over Frame Relay with Asterisk. 
 With Frame Relay work reliably with Asterisk?  Any
 experiences?

If you're talking about transporting voip calls across a path that
includes frame relay links, yes it works just fine if you frame
network is not congested.

Frame relay networks can and _may_ drop packets if the traffic exceeds
the committed information rate (cir), depending upon exactly how your
provider has their frame switches configured. Dropped packets is less
of an issue now in frame networks then what they use to be, and the
primary reason for that is the abundance of inexpensive bandwidth
currently deployed between frame switches.

There is a pecking order in terms of which packets are candidates
for being dropped, with broadcast traffic being high on that list.
It is very difficult to determine exactly where packets are dropped
as you (the user) are never notified by the frame provider when/if
they dropped any in their switches. And, if they do drop packets
you won't be able to detemine whether those that were dropped were
in fact broadcast packets or tcp/udp packets, etc. The BECN and 
FECN counts can be used to determine if the frame provider is 
recognizing whether you exceeded your cir rate, however in most 
real world implementations a BECN or FECN does _not_ translate 
into a dropped packet (at least in the US).

You might want to download Qcheck (it was originally written by NetIQ
but spun off to another company now) to evaluate the end-to-end
bandwidth. Its a free utility that will help determine what is
actually available in terms of bandwidth.

If your frame network is congested, you might be able to implement
QoS at the border routers to give some preference to voip packets.

Rich


___
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] Voice over Frame Relay Asterisk

2005-03-05 Thread asterisk phones
Great, thanks, that was the information I was looking
for.


--- Rich Adamson [EMAIL PROTECTED] wrote:

  Has anyone done Voice Over Frame Relay with
 Asterisk. 
  With Frame Relay work reliably with Asterisk?  Any
  experiences?
 
 If you're talking about transporting voip calls
 across a path that
 includes frame relay links, yes it works just fine
 if you frame
 network is not congested.
 
 Frame relay networks can and _may_ drop packets if
 the traffic exceeds
 the committed information rate (cir), depending upon
 exactly how your
 provider has their frame switches configured.
 Dropped packets is less
 of an issue now in frame networks then what they use
 to be, and the
 primary reason for that is the abundance of
 inexpensive bandwidth
 currently deployed between frame switches.
 
 There is a pecking order in terms of which packets
 are candidates
 for being dropped, with broadcast traffic being high
 on that list.
 It is very difficult to determine exactly where
 packets are dropped
 as you (the user) are never notified by the frame
 provider when/if
 they dropped any in their switches. And, if they do
 drop packets
 you won't be able to detemine whether those that
 were dropped were
 in fact broadcast packets or tcp/udp packets, etc.
 The BECN and 
 FECN counts can be used to determine if the frame
 provider is 
 recognizing whether you exceeded your cir rate,
 however in most 
 real world implementations a BECN or FECN does _not_
 translate 
 into a dropped packet (at least in the US).
 
 You might want to download Qcheck (it was originally
 written by NetIQ
 but spun off to another company now) to evaluate the
 end-to-end
 bandwidth. Its a free utility that will help
 determine what is
 actually available in terms of bandwidth.
 
 If your frame network is congested, you might be
 able to implement
 QoS at the border routers to give some preference to
 voip packets.
 
 Rich
 
 
 ___
 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
 





__ 
Celebrate Yahoo!'s 10th Birthday! 
Yahoo! Netrospective: 100 Moments of the Web 
http://birthday.yahoo.com/netrospective/
___
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] Voice over Frame Relay Asterisk

2005-03-04 Thread asterisk phones
Has anyone done Voice Over Frame Relay with Asterisk. 
With Frame Relay work reliably with Asterisk?  Any
experiences?





__ 
Celebrate Yahoo!'s 10th Birthday! 
Yahoo! Netrospective: 100 Moments of the Web 
http://birthday.yahoo.com/netrospective/
___
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] Voice over Frame Relay Asterisk

2005-03-04 Thread Steven Critchfield
On Fri, 2005-03-04 at 12:44 -0800, asterisk phones wrote:
 Has anyone done Voice Over Frame Relay with Asterisk. 
 With Frame Relay work reliably with Asterisk?  Any
 experiences?

Doesn't look like you visited google first. Nor did you bother to look
at the code.

channels/adtranvofr.h

[EMAIL PROTECTED]:/usr/src/asterisk/configs$ more adtranvofr.conf.sample 
;
; Voice over Frame Relay (Adtran style)
;
; Configuration file

[interfaces]
;
; Default language
;
;language=en
;
; Lines for which we are the user termination.  They accept incoming
; and outgoing calls.  We use the default context on the first 8 lines
; used by internal phones.
;
context=default
;user = voice00
;user = voice01

-- 
Steven Critchfield [EMAIL PROTECTED]

___
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