We running a PBX with Asterisk 1.8 and have a strange issue. On the moment we receive a call by a SIP trunk from one of our providers we notice that other channels quality is decreasing.
We even measured it by sniffing the RTP streams on the server and noticed that some of the packages get delayed on the moment an incoming call is setup. We increased the jitter buffer from 50 to 150 but it didn’t help.
Have others maybe ideas what we can do or analyse to get this issue solved.
measure your bandwidth every sip channel will consume bandwidth so make the math to know if you can handle more than one call. Create QoS and use a better codec.
[quote]We even measured it by sniffing the RTP streams on the server and noticed that some of the packages get delayed on the moment an incoming call is setup.[/quote] This issue, is related with network latency and poor or none Qos. And as navaismo said try to use narrow-band codec like G729. License is needed for this codec.
Thanks for you reply.
We have enough bandwidth 100mb and we using a QoS of 5. The server has not a huge load, 10-14 channels running on the same time.
I think I descibed my problem poorly. Every time a incomming call is coming in. Outgoing RTP packages are getting delayed. We measured this on the server itself. On the point of measurement the packeges didn’t went on the network yet.
Is that 100 milli-bits/second, per minute or per hour? I guess you meant per nano-second, i.e. 100 Mb/s
Are you, by any chance, using a virtual machine, as this could be the result virtual memory thrashing as the result of running on an under-resourced VM?
I forgot the s, 100 mb/s i meant. And no we not running on a virtual machine.
Its a 12 processor Intel® Xeon® CPU X5650 @ 2.67GHz. 32Gb mem and 500 GB HD. My feeling is that the hardware is not the issue. But we have no clue what the issue can be.