Voice choppy - but only in one direction


#1

I’m having trouble with a termination service provider and trying to figure out what the problem is. The problem is that I’m getting choppy audio, but only in one direction.

  1. SIP to SIP calls are perfect.
  2. Incomming calls are fine.
  3. Outgoing calls - I can here them fine - but I’m choppy to them.

I have a friend using the same service and he has his own asterisk box and he’s having exactly the same problem.

We got a different termination service and both of us can make outgoing calls without the choppy voice. But - I’m trying to figure out the problem for my first provider who is running Asterisk servers.

So - the real question is - what is the first provider doing wrong - or is there something about our servers that might be incompatible with his servers.

Tried several codecs - no difference.
Tried SIP and IAX2 - no difference.

Both servers are in a good colo and have plenty of bandwith - no NAT.

What would make audio bad consistently only in one direction?


#2

did you try pings in both directions and do a traceroute in both directions, to determine what sort of packet loss, and variation you get and determine if the routes are the same in both directions (meaning you’ll need the provider to help on this one do the pings and traceroutes back to you).

p


#3

[quote=“p_lindheimer”]did you try pings in both directions and do a traceroute in both directions, to determine what sort of packet loss, and variation you get and determine if the routes are the same in both directions (meaning you’ll need the provider to help on this one do the pings and traceroutes back to you).

p[/quote]

There’s no packey loss involved. It really has us stumped.


#4

what about the variance - and did you have the supplier try in your direction? There’s more than packet loss that will kill the quality.

p


#5

[quote=“p_lindheimer”]what about the variance - and did you have the supplier try in your direction? There’s more than packet loss that will kill the quality.

p[/quote]

I will check into that but what is “variance”?


#6

it translates into jitter. Let ping run for a while and watch the results. Are they all very consistent or are they all over the map. (And do it at a time when you know you are having call problems). I had one DSL line that at times had fairly reglular pings of 44-48msec, but at other times they were 44-120msec with a lot of randomness. The other issue, these ping times were to the ISP primary gateway (one hop away, very close).
Should have been more like 10msec. After a couple of hours with the provider and getting them to make some changes, we got it to 8msec with normal behavior. (typically 8-9msec with normal behavior not going much above 11-12msec).

p


#7

What phones are you using? Is your rtp packet size at 0.020?

This is a common problem in Sipura phones


#8

[quote=“plummy”]What phones are you using? Is your rtp packet size at 0.020?

This is a common problem in Sipura phones[/quote]

I’m using a linksys pap2-n2 - not sure what the packet size is. What should it be and why?


#9

some of the Siupra ATA’s I’ve seen set the packet to 0.03 by default for some strange reason. Go tot he sip tab in admin/advanced and check the RTP Packet Size. You want that at 0.020 typically.

p


#10

You should also check to see if the phone endpoint you are using isn’t set to do “silence suppression”.

Asterisk doesn’t deal with silence suppression very well at all. You should be transmitting an audio stream all the time, even if that stream is just silence.