Jitter Buffer
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I am relaying a T1 over ethernet, using an Adtran MX408e.
Since it is a TDM interface packets must be received prior to the jitter buffer on the MX408e exhausting. The MX408e only has a max jitter buffer of 100ms. The jitter on my present connection is ~300ms.
Has someone done this with traffic shaping or dummynet? I have been playing with queues to no avail.
It is not VOIP traffic so a short delay will not pose any problems.
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I'm not entirely privy to T1 terminology, so I may make some incorrect assumptions.
My question is do you know why the jitter is so incredibly high? T1s are rated for 1-5ms of jitter and Ethernet is typically in the microseconds. You're talking about latencies 100x greater that normal. While the max buffer may be 100ms, you shouldn't be filling a buffer on a synchronous connection.
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Are you relaying a T1 over "ethernet" or ethernet then the internet. A good switch should be able to forward frames with latency measured in microseconds, not 300 milliseconds.
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The set up looks like this:
+–----+ +--------+
T1+--+MX408e++ETHERNET+-+PFSENSE |+----+
+------+ +--------+ |
|
MOBILE CONNECTION/OPENVPN |
+----------------------------------------+
|
|
| +-------+ +------+
+-+|PFSENSE++ETHERNET++MX408e+--+T1
+-------+ +------+The T1s are either end are used for relaying telemetry data from two arcane pieces of equipment between power plants a delay of under 2 seconds end to end is acceptable.
The jitter comes when the connection is piped over the mobile connection. Asterisk and FreeSwitch seemed to have some jitter buffering capabilities, but I am not sure how to implement..
The buffer would work like this (assume 1 packet per millisecond):
-Wait and capture 300 packets–->Send 1 packet every millisecond. -
I might be confused but wouldn't the 100ms buffer on the mx408e always be the limiting factor in the above scenario?
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I think you're going to have a hell of a time trying to get a TDM circuit transported over a 300ms RTT VPN link. No amount of traffic shaping is going to make that any better. SIP trunking would probably be better, but 300ms is a lot for that too.
