[cisco-voip] Optimizing RTP on the WAN.

[Joe Pollere (US)]

Sorry that should read "will be exaggerated beause of the larger payload size."

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From: cisco-voip-bounces at puck.nether.net on behalf of Joe Pollere (US)
Sent: Fri 11/10/2006 4:07 PM
To: Scott ODonnell; cisco-voip at 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/article.asp?p=357102&seqNum=1&rl=1
 
<snip>

Table 2-1 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 at puck.nether.net on behalf of Scott ODonnell
Sent: Fri 11/10/2006 3:55 PM
To: cisco-voip at 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 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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