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ftarz at mindspring.com
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 Posted: Thu May 01, 2008 3:19 pm Post subject: [asterisk-users] Minimum upload speed for Asterisk? |
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I'm running an Asterisk box that's connected to the world via 5MB
down/384kB up cable internet service. I've noticed that the sound
quality for both IAX and SIP calls sometimes starts to suffer. IVR
prompts and MOH frequently have slight pauses from the outside, but
sound fine from inside calls.
Is 384kB up too slow?
Is there any guidance for the minimum upload speed for an Asterisk box?
Frank |
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erikerik at gmail.com
Guest
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| Posted: Thu May 01, 2008 3:33 pm Post subject: [asterisk-users] Minimum upload speed for Asterisk? |
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On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski <ftarz at mindspring.com> wrote:
| Quote: |
Is 384kB up too slow?
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Probably not.
| Quote: | Is there any guidance for the minimum upload speed for an Asterisk box?
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I'm guessing this is for just a few calls at a time, correct? I'd
guess that rather than these quality issues being caused by cramped
bandwidth, they're actually being caused by latency issues. Have you
ever checked the latency of the connection between your asterisk
server and your SIP/IAX endpoint? If it's really high (say 300ms+) or
if the latency is really erratic, you'll have quality issues.
You didn't mention whether you are doing traffic shaping on your
upstream connection, so I'll assume you're not. That would be
something good to look into - with traffic shaping, you can prioritize
your VoIP traffic over all other types of network traffic.
-erik |
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bwentdg at pipeline.com
Guest
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| Posted: Thu May 01, 2008 4:23 pm Post subject: [asterisk-users] Minimum upload speed for Asterisk? |
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You also need to check for Packet Loss on the Link
Erik Anderson wrote:
| Quote: | On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski <ftarz at mindspring.com> wrote:
| Quote: | Is 384kB up too slow?
|
Probably not.
| Quote: | Is there any guidance for the minimum upload speed for an Asterisk box?
|
I'm guessing this is for just a few calls at a time, correct? I'd
guess that rather than these quality issues being caused by cramped
bandwidth, they're actually being caused by latency issues. Have you
ever checked the latency of the connection between your asterisk
server and your SIP/IAX endpoint? If it's really high (say 300ms+) or
if the latency is really erratic, you'll have quality issues.
You didn't mention whether you are doing traffic shaping on your
upstream connection, so I'll assume you're not. That would be
something good to look into - with traffic shaping, you can prioritize
your VoIP traffic over all other types of network traffic.
-erik
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mgraves at mstvp.com
Guest
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| Posted: Thu May 01, 2008 9:03 pm Post subject: [asterisk-users] Minimum upload speed for Asterisk? |
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On Thu, 01 May 2008 16:19:26 -0400, Frank Tarczynski wrote:
| Quote: | I'm running an Asterisk box that's connected to the world via 5MB
down/384kB up cable internet service. I've noticed that the sound
quality for both IAX and SIP calls sometimes starts to suffer. IVR
prompts and MOH frequently have slight pauses from the outside, but
sound fine from inside calls.
Is 384kB up too slow?
|
No. But what you can do will depend upon codec selection, QoS & traffic
shaping in your router.
| Quote: | Is there any guidance for the minimum upload speed for an Asterisk box?
|
Rough guidance:
G.711 needs approx 80 kbps per call leg
G.729a needs approx 32 kbps per call leg
See www.voip-info.org and search for codecs.
Michael
--
Michael Graves
mgraves<at>mstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mjgraves at pixelpower.onsip.com
skype mjgraves
54245 at fwd.pulver.com |
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stotaro at totarotechn...
Guest
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| Posted: Fri May 02, 2008 7:41 am Post subject: [asterisk-users] Minimum upload speed for Asterisk? |
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On Thu, May 1, 2008 at 10:03 PM, Michael Graves <mgraves at mstvp.com> wrote:
| Quote: | On Thu, 01 May 2008 16:19:26 -0400, Frank Tarczynski wrote:
| Quote: | I'm running an Asterisk box that's connected to the world via 5MB
down/384kB up cable internet service. I've noticed that the sound
quality for both IAX and SIP calls sometimes starts to suffer. IVR
prompts and MOH frequently have slight pauses from the outside, but
sound fine from inside calls.
Is 384kB up too slow?
|
No. But what you can do will depend upon codec selection, QoS & traffic
shaping in your router.
| Quote: | Is there any guidance for the minimum upload speed for an Asterisk box?
|
Rough guidance:
G.711 needs approx 80 kbps per call leg
G.729a needs approx 32 kbps per call leg
See www.voip-info.org and search for codecs.
Michael
--
Michael Graves
mgraves<at>mstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mjgraves at pixelpower.onsip.com
skype mjgraves
54245 at fwd.pulver.com
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Try the same calls with the other devices on your network turned off
or unplugged from the network. All it takes is a semi large or even
small but bandwidth intensive file transfer (bit torrent or FTP for
example) to mangle your audio.
If the audio is fine after limiting to only VoIP traffic then you can
figure out the best way to prevent bandwidth hogging, whether it be to
throttle the offending application or make rules on your router.
Thanks,
Steve Totaro |
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jfaubion at tx.rr.com
Guest
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| Posted: Fri May 02, 2008 8:33 am Post subject: [asterisk-users] Minimum upload speed for Asterisk? |
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| Quote: | to suffer. IVR prompts and MOH frequently have slight pauses
from the outside, but sound fine from inside calls.
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If the pauses your talking about are at the beginning of the call but are
fine after the call is answered, I'd suspect latency issues. If the pauses
are no longer than just glitches, static or pops then suspect jitter. If the
pauses are longer, and affect IVR, MOH and voice traffic equally, then look
for packet loss. If the voice traffic is fine but the IVR and MOH are
affected, look for I/O bottlenecks in your hardware. Try using My Traceroute
(mtr) to your SIP provider. This may point out packet loss and latency
issues. I say may because ICMP traffic could be handled differently than
SIP/RTP traffic.
| Quote: | Is 384kB up too slow?
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Depending on the codec used, this should be sufficient for 4 or more
simultaneous calls as long as the latency, packet loss and jitter don't kill
it.
John Faubion |
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