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Alinux
Joined: Jan 30, 2008
Posts: 14
Status: Offline
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Aug 28, 2008 - 08:42 AM |
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Hi All,
I'm using A2billing application in order to make callback calls through my asterisk server...Everything looks fine except the voice quality...There is a lot of noise in the call with different codecs(G711, and G729)...Please note that when I do make a call to a PSTN line from a softphone all is fine.
I used several carriers with the same result...What do you suggest?
Regards |
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ianplain
Site Admin
Joined: Jul 05, 2004
Posts: 3011
Location: Bath UK
Status: Offline
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Aug 28, 2008 - 09:06 AM |
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Hi
Is the calls in and out using Voip / PSTN or what. If the in and out leg are Voip then its more than likely to be a bandwidth issue .
Ian |
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Alinux
Joined: Jan 30, 2008
Posts: 14
Status: Offline
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Aug 28, 2008 - 09:14 AM |
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Hi
The calls are placed using a web interface, that web interface is on the same server as the asterisk server. After that the call is setup on both logs in the following fashion:
PSTN <------->Asterisk<------->PSTN
The asterisk server itself has a lot of free bandwidth, I have checked my MRTG graphs and using iftop.
Regards |
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ianplain
Site Admin
Joined: Jul 05, 2004
Posts: 3011
Location: Bath UK
Status: Offline
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Aug 28, 2008 - 09:19 AM |
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Hi
Are the calls to the PSTN Voip or using TDM trunks? If they are voip what is the bandwidth of the link?
If you are bridging TDM what card are you using ?
Ian |
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Alinux
Joined: Jan 30, 2008
Posts: 14
Status: Offline
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Aug 28, 2008 - 09:42 AM |
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Hi
Thanks again for the help. That is PSTN VOIP and the link bandwidth is a dedicated 10 Megs out of which roughly 1 megs are used.
Thanks |
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ianplain
Site Admin
Joined: Jul 05, 2004
Posts: 3011
Location: Bath UK
Status: Offline
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Aug 28, 2008 - 09:55 AM |
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Hi
Are the trunks IAX or SIP ? also how many calls before you get the problem ? If the usage is 1 meg then I guess its 12 calls in total, Is that correct?
Ian |
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Alinux
Joined: Jan 30, 2008
Posts: 14
Status: Offline
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Aug 28, 2008 - 10:31 AM |
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Hi
Those are SIP links. Apart from that the usage is for other services on other servers for the same link. The server itself has not other calls, this is a test server and we are testing callback quality on it. The sound quality is really bad.
I.e.
Server A ----\
Server B -----| -> Cisco Router with 10 MB Link
Server C ----/ |
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ianplain
Site Admin
Joined: Jul 05, 2004
Posts: 3011
Location: Bath UK
Status: Offline
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Aug 28, 2008 - 11:36 AM |
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Hi
The quality issues will be with your network, you need to look at that and work out whats going on.
Ian |
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Alinux
Joined: Jan 30, 2008
Posts: 14
Status: Offline
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Aug 28, 2008 - 11:39 AM |
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Hi
Thanks for the hint, however I would like to point out that calls using a sip phone to PSTN are great...does this include anything ..? Or maybe not ? |
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ianplain
Site Admin
Joined: Jul 05, 2004
Posts: 3011
Location: Bath UK
Status: Offline
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Aug 28, 2008 - 12:22 PM |
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Hi
You may find that the problem is at the router.
you need to sniff the network and look at the rtp packets.
The server is a dedicated server not a VM ?
Ian |
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Alinux
Joined: Jan 30, 2008
Posts: 14
Status: Offline
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Aug 28, 2008 - 12:41 PM |
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Hi this is a dedicated server. What exactly should I be looking for with regard to RTP?
Thanks |
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