I am using Asterisk to help promote a music artist by letting callers hear various songs by selecting them from the keypad (Press ‘1’ to hear this song, Press ‘2’ to hear that song, etc.). Unfortunately, the sound quailty that callers hear is very poor. I tried installing a high quality soundcard, but that didn’t help at all.
I have also tried playing the music as .WAV files (8Khz, mono) using the Playback() application as well as .MP3 files using the MP3Player() application, but the sound quality was the same.
Now, this is not-so-powerful ‘test’ system (Pentium III, 866MHz, 256 RAM). Would improving the processor and RAM increase the sound quality?
The volume on the music is fine, I’m just trying to find a way to make it sound clearer. I realize that the PSTN frequency range is limited to 400Hz to 4400Hz, but I know the sound can be more crisp than what I’m getting.
Thanks in advance for your help!
i would re-encode the full-quality files into raw format, so that asterisk doesn’t have to transcode twice…it can transcode directly from raw to whatever format the user is capable of (most likely ulaw, if it’s a PSTN call)
are you currently testing over a POTS line? if so, you might test with a local SIP phone, as that would probably provide better quality and allow you to rule out asterisk as the problem.
our MOH is exceptionally clear over both SIP and PSTN links, but we are also utilizing T1 lines which should have better sound quality than a standard 2 wire analog line.
hope this helps.
Well, I think I have ruled out Asterisk as the problem. I set up an extension to go directly to Music On Hold to test from both a SoftPhone client (Xten Lite) on my Asterisk server and simultaneously called from a PSTN line. The sound quality was great over the SIP phone, but still sounded bad over my PSTN call. I also tested using a GIZMO SIP call which actually takes me out over the WAN and back into my Asterisk server and I got the same poor sound quality.
So with that being said, I’m assuming that the problem is with my VoIP provider or my actual WAN connection. I really don’t think that it’s my WAN connection because I’m getting at least 384K upstream and uLaw only uses 56 or 64Kbps per connection, correct?
Any thoughts or experience with a VoIP provider’s network being the cause of poor sound quality for PSTN calls?
ulaw uses 64kbps per LEG of the call (you have both an inbound and outbound leg for each duplex call) plus overhead, so it’s more like 160kbps.
another thing to look at would be the latency and/or jitter of the voice packets, as UDP doesn’t have all of the controls TCP does, but has much less overhead as well.
you MIGHT try switching to GSM as a codec - it’s about 80% as good as ulaw but only 13kbps (i think) for each leg…it MIGHT help.
Thanks for your help!
I’ve tried GSM and still getting the same results. The only thing I can possibly think of is either my VoIP provider’s network OR my Internet connection (DSL - 1.3 Mbps Down/384 Kbps Up).
I found the same issue, but with the streaming music I get great sound…
So, I setup a shoutcast server with a wk2 pro as the “media storage server” for the stream,
(look at andromeda by turnstyle.com/andromeda/ )
Create a custom radio station so to speak…
maybe you can do something like that…
only other thing would be to check the latency/jitter/packet loss between you and your provider - cable and DSL aren’t exactly low-latency pipes, and it might just be that your line is causing jitter and/or packet loss.
eyebeam (and possibly xlite) has a basic call statistics popup which tells you the MOS and jitter stats for that particular call - it might help you to determine if that is what’s happening…
good luck, i HATE troubleshooting this crap.
Another place you might look is in your codecs.conf file to see what your sound quality and complexity are set at. If you are only looking at good quality coming out of the server, you can set those high. This will cause latency for duplex calling, but for just listening to music it should work great! Hope this helps.
Thanks for all of your help. I have finally found the problem and it’s with my VoIP provider. I set up a free GIZMO account and registered my Asterisk server as a client so that I could call in and test sound quality going over another VoIP provider’s network. Sure enough, the sound quality was clear and undistorted. I had my VoIP provider send my calls through another gateway and I still got the same result.
Now I need to find another VoIP provider who can provide:
- Good sound quality on calls.
- Flat rate for unlimited inbound calls.
- Multiple channels that can be shared across a range of DIDs to allow many simultaneous inbound calls.
- DIDs from a variety of US cities (and preferably international numbers too).
Any recommendations would be greatly appreciated. My current VoIP provider offers all of the above EXCEPT the good sound quality. Otherwise, I’d stay with them.