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Unread 05-17-2007, 05:10 PM   #21
apoermandya
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Hi,

Voice streaming data needs a guaranteed bandwidth available to ensure a good quality conversation.
G.711 codec (64 kbps voice bandwidth) would require 84 kbps of network bandwidth while G.729 codec (8 kbps) would require about 24 kbps network bandwidth.
I don't have any experience with GSM but I think the voice bandwidth is almost the same with G.729, around 9.6kbps.

The additional bandwidth is required for the extra RTP (Real Time Protocol) header for every voice packets.

VoIP Quality is affected by delay and jitter.
Here is a link that explains about VoIP Quality of Service

The break up is most likely because of the ISP in China does not have QoS setup for voice traffic.

My suggestion is to use low bandwidth codec like G.729 or GSM to minimise the bandwidth requirement.

Have you tried to call using a direct IP Address by dialling
sip:xxxxxx@A.B.C.D
where
xxxxxx is your Voxalot number
A.B.C.D is your IP Address from your ISP

If you use a router, you may need to setup the port forwarding to forward TCP 5060 to your PC or ATA adapter.

Just my 2 cents.
Hope this helps.

Adrian
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Unread 05-17-2007, 05:48 PM   #22
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Most likely the voice quality that your friend experiencing is due to "jitter".

Here is some information regarding
VoIP jitter


Voice streaming data needs a guaranteed bandwidth with a constant latency/delay to avoid jitter.

G.711 codec (64 kbps voice bandwidth) would require 83 kbps network bandwidth.
G.729 codec (8 kbps voice bandwidth) would require 27 kbps network bandwidth.
GSM codec (13 kbps voice bandwidth) would require 29 kbps network bandwidth.

Reference :
Cisco.com: Voice Over IP - Per Call Bandwidth Consumption

The additional bandwidth is because of there is an additional RTP (Real Time Protocol) header for every voice packets that traverse on the TCP/IP networks.

The ISP in Shanghai may not have a QoS setup for voice traffic. That's why even you have plenty of download/upload bandwidth, the voice quality is bad.

I suggest you use G.729 or GSM codec to minimise the bandwidth requirement.

Just my 2 cents. Hope this helps.

Adrian
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Unread 05-17-2007, 11:53 PM   #23
vk4akp
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Hi, not sure what is going on here as I received a reply in the email but it's not listed here.

But thanks to > apoermandya < I received your reply and it is very helpful..

Yes I think basically the lack of decent QOS at her location is probably a big issue.

Some days it works good. SOme other days it's crap.

She lives in a big hirise so the internet there is probably open to all sorts of issues.

Anyhow, thanks for all the interesting info on codec's and the bandwidth used.

.-.
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Unread 05-18-2007, 12:34 AM   #24
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apoermandya's posts went into the moderate queue as they contained links and apoermandya is new user (spam protection).

I have subsequently approved the posts and they should now be visible.
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Martin

Please post support questions on the forum. Do not send PMs unless requested.
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Unread 05-18-2007, 01:07 AM   #25
vk4akp
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Question Dial Up over SIP????

Quote:
Originally Posted by martin View Post
apoermandya's posts went into the moderate queue as they contained links and apoermandya is new user (spam protection).

I have subsequently approved the posts and they should now be visible.
OK, Thanks Martin.

Hey I have a really whacky question for your.

In theory is it possible to use a dial up modem to access a data link through SIP?

For example.

A person sets up a dial up modem to ring a SIP gateway and then enter the correct code so it passes through to my system.

If I had some sort of software set up to pass the connection to a modem whould this work? Even if it was at a slow rate?

Also would I have to use a fax codec?

And is there some sort of setup to acheive this at the receive end via software only?

.-.-.
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