Guys,
I’m slowly getting sick of call quality issues with asterisk, even though it happens once every week or something due to latency on the internet bouncing up and down, it undermines my companies reputation.
Is there a way to prevent a call being affected by the quality of the internet?
something more then just “yes speak to your ISP” because internet will always bounce up and down, is there something can do to the way asterisk handles that.
It depends of course on where the congestion lies, but if you suspect your local network, you can see if your router supports QoS such as Differentiated Services, and set the bits in Asterisk packets appropriately. If it’s beyond your network…
Ian
Hi
Firstly, The primary answer will be speak to or move ISP. That being said have you got jitterbuffers setup OK ? and are you sure your have the upstream bandwidth. Also look for an ISP that is linked to voip supplier.
It may also be a good idea to do a traceroute to your ITSP and see how many hops are involved.
What you mustn’t forget that it isnt Asterisk’s fault that you have a poor quality DSL line.
Ian
defenitly not asterisk’s fault, and i havent forgotton to mention im sick of the line deciding the quality either…
can someone point me in the right direction of jitter buffering ive tried it before on many of its levels and it made things worse…
i have tried rtp packetization to 10ms which helped a little.
anyone know how i can confirm the packetization change did take affect?
anyone know how to change packetization on cisco 7940 from 10ms to 20ms to suit asterisk.?
anyone know another slightly more active asterisk forum i can also try?
Hi
what do you mean by [quote]and i havent forgotton to mention im sick of the line deciding the quality either… [/quote] ?
and with respect to jitterbuffers see voip-info.org/wiki/view/Aste … buffer.txt
finaly checking packet size, and general trouble shooting use tcpdump to cature th epackets and use wireshark.
Ian
[quote=“rnbguy”]anyone know how i can confirm the packetization change did take affect?
anyone know how to change packetization on cisco 7940 from 10ms to 20ms to suit asterisk.?
anyone know another slightly more active asterisk forum i can also try?[/quote]
doesn’t get much more active than this for asterisk… you basically asked an impossible question to answer.
if your ISP sucks to the point that your IP calls suck… there is not much that asterisk can do, so I am not surprised you didn’t get a huge response.
that’s an example of someone who doesnt know what their speaking about, ive had a few suggestions to help on trixbox forum, people like you kill threads too quick… keep it to yourself.
im using this to help when we use sip trunks:
[macro-hangupcall]
exten => s,1,NoOp(${CHANNEL(rtpqos)})
to find the issue (packet loss counts etc)
however what can i use for iax2? i know “iax2 show netstats” is good but how do you call that from a dialplan on call hangup to log call stats.
Perhaps if you started off with something more like:
I’m slowly getting sick of call quality issues with MY CURRENT ISP, even though it happens once every week or something due to latency on the internet bouncing up and down, it undermines my companies reputation. Is there something Asterisk can do to help me here?
You might get more help. Your original statement leads people to believe that you don’t understand the technology you are working with… but you sound like you think you do.
I am just saying, come to the forums with a better attitude and you will get more help. Please look up my post history, you will see that I am a very active member who is always willing to help people who act with respect toward the community.
if you are really serious about ending call quality issues over the internet… you may need to look into a managed Service such as MPLS that can assure Qos / COS is maintained from your site to the ITSP site…
typically the internet latency is more than sufficient for data transmission but for UDP voice calls any latency can cause issues…
-Christopher
your suggestion fell in time with that of my ISP, they have suggested to place us on MPLS to end latency issues.