Hi everyone.
We need to specify the following codec setting under the host definition:
allow=g729:60
We need to run on 60ms payload sizes due to the cost of bandwidth in our country. The rest of the network is already running at 60ms and works perfectly. We need the hosted pbx on this as well.
Is there a way we could add an additional codec to the Thirdlane interface that is simply g729:60?
I have tried using the "other options" field but it does not return the option next time you open the page. It also only works if you remove all other codecs from the list.
Would appreciate any help?
Thanks
Heinrich
Hi Thanks for the reply
Hi
Thanks for the reply Erik.
Yeah we have really made some great advancements (commercially) with the 60ms payload size in our country.
To answer your question: Yes I am familiar with the advantages of IAX and we will probably be implementing infrastructure for the Asterisk guys soon. Unfortunately it's not really an option for the hosted pbx where we only have phones in the field.
Is there somewhere I can simply modify the interface to put "g729:60" in the config files instead of "g729"? Otherwise I should be mentioning this in the new features forum?
Thanks
Heinrich
i cant get a definitive
i cant get a definitive answer on codecs.conf as to whether you can define parameters of any codec there or just a select few. If its the former, then it should solve your issues.
Hi Erik. Thanks for the
Hi Erik.
Thanks for the feedback. I have setup the global codec definition for g729:60 (in sip.conf). And it is working for now.
I had to remove all the codec definitions for each extension and trunk for this to work. This is obviously taking away the functionality to define codecs per provider / extension.
Should I go ahead and request this feature in the thirdlane Gui?
Thanks again
Heinrich
the way you'll be
the way you'll be implementing it in the gui is in the works, but not an immediate fix. Its not going to be there in a way for everyone to see, it will cause a lot of problems for everyone else if they accidentally select this one instead of the standard 20ms and most people are not adept enough to understand the difference. What will be arriving is the ability to add more customization to the screens so that you can put it in your GUI without impacting others.
Wonderful!
Hi Erik
That is great news, thanks!
Is it simply going to be a software upgrade of the gui?
Another thing I came across is the recording quality with the 60ms payload size. Its extremely bad. Almost sounds like its only recording 20ms of the 60ms each time. Any ideas on this? I cant seem to find anything on other asterisk forums.
Mixmonitor is used by thirdlane, so i was thinking of using Monitor that does not decode - mix - encode the audio streams. I could mix the files at a later stage. Do you think this is worth trying?
Thanks
Heinrich
Damn, you drop one packet and you'd definitely know it. Packet loss concealment isn't going to work on such a big gap, thats for sure :-). I'll look into what can be done, but for now it looks like you're stuck manually editing the files. How expensive is the bandwidth that the headers alone are cutting into your cost?
Not that this would work for your phones, but have you ever read up on IAX2 trunking? Works really well to pay the payload cost once and stack multiple calls on it. Granted, there is no provisions for t38 in iax2, but its a huge bandwidth saver, just make sure your timing source is working.