[asterisk-users] Minimum upload speed for Asterisk?
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimum upload speed for Asterisk?
On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski <[EMAIL PROTECTED]> wrote: > > Is 384kB up too slow? Probably not. > 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimum upload speed for Asterisk?
You also need to check for Packet Loss on the Link Erik Anderson wrote: > On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski <[EMAIL PROTECTED]> wrote: > >> Is 384kB up too slow? >> > > Probably not. > > >> 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 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimum upload speed for Asterisk?
On Thu, 01 May 2008 16:19:26 -0400, Frank Tarczynski wrote: >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. >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 mgravesmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimum upload speed for Asterisk?
On Thu, May 1, 2008 at 10:03 PM, Michael Graves <[EMAIL PROTECTED]> wrote: > On Thu, 01 May 2008 16:19:26 -0400, Frank Tarczynski wrote: > > >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. > > > >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 > mgravesmstvp.com > http://blog.mgraves.org > o713-861-4005 > c713-201-1262 > sip:[EMAIL PROTECTED] > skype mjgraves > [EMAIL PROTECTED] 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Minimum upload speed for Asterisk?
> to suffer. IVR prompts and MOH frequently have slight pauses > from the outside, but sound fine from inside calls. 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. > Is 384kB up too slow? 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
