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Home > Archive > Voice over IP Cisco > November 2006 > Optimizing RTP on the WAN.
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Optimizing RTP on the WAN.
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| Scott ODonnell 2006-11-13, 8:10 am |
| I have a customer that is getting clobbered high WAN utilitization due to
RTP streams.
While I recognize that at some point you just need more bandwidth are there
any other ways to reduce the bandwidth RTP requires beyond codec selection
and CRTP?
I vaguely remember back in the old "toll bypass" days, you could adjust the
number of samples per packet or the sample rate and it could make a real
difference in the bandwidth.
Any knobs like that in CallManager?
Scott
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| Joe Pollere \(US\) 2006-11-13, 8:10 am |
| You could change the default sampling rate from 20 ms to 30 ms but the bandwidth savings is really not that significant. However If you are experiencing any dropped packets it will be exasperated beause of the larger payload size.
See this for reference:
http://www.informit.com/articles/ar...2&seqNum=1&rl=1
<snip>
Table2-1 details the bandwidth per VoIP flow (both G.711 and G.729) at a default packetization rate of 50 packets per second (pps) and at a custom packetization rate of 33 pps. This does not include Layer 2 overhead and does not take into account any possible compression schemes, such as Compressed Real-Time Transport Protocol (cRTP, discussed in detail in Chapter 7, "Link-Specific Tools").
For example, assume a G.711 VoIP codec at the default packetization rate (50 pps). A new VoIP packet is generated every 20 ms (1 second / 50 pps). The payload of each VoIP packet is 160 bytes; with the IP, UDP, and RTP headers (20 + 8 + 12 bytes, respectively) included, this packet become 200 bytes in length. Converting bits to bytes requires multiplying by 8 and yields 1600 bps per packet. When multiplied by the total number of packets per second (50 pps), this arrives at the Layer 3 bandwidth requirement for uncompressed G.711 VoIP: 80 kbps. This example calculation corresponds to the first row of Table 2-1.
Table 2-1 Voice Bandwidth (Without Layer 2 Overhead)
Bandwidth Consumption
Packetization Interval
Voice Payload in Bytes
Packets Per Second
Bandwidth Per Conversation
G.711
20 ms
160
50
80 kbps
G.711
30 ms
240
33
74 kbps
G.729A
20 ms
20
50
24 kbps
G.729A
30 ms
30
33
19 kbps
NOTE
The Service Parameters menu in cisco CallManager Administration can be used to adjust the packet rate. It is possible to configure the sampling rate above 30 ms, but this usually results in poor voice quality.
<snip>
HTH's
Joe
________________________________
From: cisco-voip-bounces@puck.nether.net on behalf of Scott ODonnell
Sent: Fri 11/10/2006 3:55 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Optimizing RTP on the WAN.
I have a customer that is getting clobbered high WAN utilitization due to RTP streams.
While I recognize that at some point you just need more bandwidth are there any other ways to reduce the bandwidth RTP requires beyond codec selection and CRTP?
I vaguely rememberback in the old "toll bypass" days, you could adjust the number of samplesper packet or the sample rate and it could make a real difference in the bandwidth.
Any knobs like that in CallManager?
Scott
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| Joe Pollere \(US\) 2006-11-13, 8:10 am |
| Sorry that should read "will be exaggerated beause of the larger payload size."
________________________________
From: cisco-voip-bounces@puck.nether.net on behalf of Joe Pollere (US)
Sent: Fri 11/10/2006 4:07 PM
To: Scott ODonnell; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Optimizing RTP on the WAN.
You could change the default sampling rate from 20 ms to 30 ms but the bandwidth savings is really not that significant. However If you are experiencing any dropped packets it will be exasperated beause of the larger payload size.
See this for reference:
http://www.informit.com/articles/ar...2&seqNum=1&rl=1
<snip>
Table 2-1 details the bandwidth per VoIP flow (both G.711 and G.729) at a default packetization rateof 50 packets per second (pps) and at a custom packetization rate of 33 pps. This does not include Layer 2 overhead and does not take into account any possible compression schemes, such as Compressed Real-Time Transport Protocol (cRTP, discussed in detail in Chapter 7, "Link-Specific Tools").
For example, assume a G.711 VoIP codec at the default packetizationrate (50 pps). A new VoIP packet is generated every 20 ms (1 second / 50pps). The payload of each VoIP packet is 160 bytes; with the IP, UDP, and RTP headers (20 + 8 + 12 bytes, respectively) included, this packet become 200 bytes in length. Converting bits to bytes requires multiplying by 8and yields 1600 bps per packet. When multiplied by the total number of packets per second (50 pps), this arrives at the Layer 3 bandwidth requirement for uncompressed G.711 VoIP: 80 kbps. This example calculation corresponds to the first row of Table 2-1.
Table 2-1 Voice Bandwidth (Without Layer 2 Overhead)
Bandwidth Consumption
Packetization Interval
Voice Payload in Bytes
Packets Per Second
Bandwidth Per Conversation
G.711
20 ms
160
50
80 kbps
G.711
30 ms
240
33
74 kbps
G.729A
20 ms
20
50
24 kbps
G.729A
30 ms
30
33
19 kbps
NOTE
The Service Parameters menu in cisco CallManager Administration can be used to adjust the packet rate. It is possible to configure the sampling rate above 30 ms, but this usually results in poor voice quality.
<snip>
HTH's
Joe
________________________________
From: cisco-voip-bounces@puck.nether.net on behalf of Scott ODonnell
Sent: Fri 11/10/2006 3:55 PM
To: cisco-voip@puck.nether.net
Subject: [cisco-voip] Optimizing RTP on the WAN.
I have a customer that is getting clobbered high WAN utilitizationdue to RTP streams.
While I recognize that at some point you just needmore bandwidth are there any other ways to reduce the bandwidth RTP requires beyond codec selection and CRTP?
I vaguely remember back in the old "toll bypass" days, you could adjust the number of samples per packet or the sample rate and it could make a real difference in the bandwidth.
Any knobs like that in CallManager?
Scott
________________________________
Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use bythe designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contentsis strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.
-----------------------------------------
Disclaimer:
This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful. If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.
| |
| Wes Sisk 2006-11-13, 8:10 am |
| Use this and tweak to your heart's content:
http://tools.cisco.com/Support/VBC/do/CodecCalc1.do
generally only options are:
1. low bit rate codec, lower general voice quality
2. cRTP, higher router CPU utilization
3. increased packetization period/decreased packets per second, lost/
late packets impact audio stream more. also some compatibility
issues depending on how large you go. Older versions of Unity did
not support over 20msec g729. ex: CSCuk52490
/Wes
On Nov 10, 2006, at 3:55 PM, Scott ODonnell wrote:
I have a customer that is getting clobbered high WAN utilitization
due to RTP streams.
While I recognize that at some point you just need more bandwidth are
there any other ways to reduce the bandwidth RTP requires beyond
codec selection and CRTP?
I vaguely remember back in the old "toll bypass" days, you could
adjust the number of samples per packet or the sample rate and it
could make a real difference in the bandwidth.
Any knobs like that in CallManager?
Scott
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