by William Flanagan
- To follow up SIP Trunking in the previous VL, what happens if
the technology succeeds? Specifically, what are the link and router
requirements if all the branch offices rely on the IP PBX and a shared
carrier interface to the PSTN at headquarters? Two metrics stand out:
- Consider the relative size of the payload (voice information) and
the packet headers (routing and processing information).
- Full headers (Ethernet, IP, UDP, RTP) total 68 bytes, sent 50
times per second, or 27.2 kbit/s
- Encoded voice most often is either 64 kbit/s (PCM/G.711) or 8
- The total per active channel is the sum of the two, either 91.2
or 35.2 kbit/s. That's full duplex, needing the same bandwidth in both
directions. Bandwidth, while not free, usually is available at a
reasonable cost to handle as many voice calls as needed.
- All the signaling messages from branch locations will pass over
the private network (HQ-to-Branch) to reach the IP PBX at HQ. VoIP
connections routed normally will take the shortest available path. With
all the SIP trunks located at headquarters "External" calls, into and
out of the branches as well as the central site, will use the SIP
trunks. This means that bandwidth for voice traffic, not just
signaling, must be available between HQ and each branch. Some
"internal" calls between branches might take shorter paths on the
Packets Per Second
- The other possible constraint is packets per second. This number
can be more limiting than bandwidth.
- A large site could have 1000 calls up at once, generating 50 x
1000 = 50,000 PPS. A gateway router that carries unified communications
needs to deal with data and probably video as well as voice, which can
push the PPS reading into the millions. At headquarters you can justify
big hardware to deal with big traffic. For example, Cisco's Switch
Fabric Module 2 is said to allow designing for 30 million packets per
- Remote sites, if they are numerous, present a different problem.
The access device that's adequate for bit rate may not have the
horsepower for the necessary packet rate. For comparison, Cisco's 1900
series branch router is considered very busy at 5,000 PPS and may under
some configurations start dropping packets if the multicast rate
reaches 100 PPS--as little as two conference calls on multicast.
- Upgrading thousands (or even hundreds) of devices may need more
CapEx than available.
- Data compression as used in WAN acceleration doesn't work well
for voice information--it's too random. But the headers in voice
packets are far from random--addresses remain the same during a call
and certain fields increment in predictable ways. So header compression
is very practical. As little as one byte can replace all of the IP,
UDP, and RTP headers. A link-layer header (Layer 2 Ethernet for
example) is still required. Routers can apply one of several standard
methods for header compression on a link between adjacent routers.
Paths longer than one hop require the same configuration of every
intermediate router to process packets at each input and output port.
- Frame Relay Implementation Agreement 11 (FRF.11) defined (in
1996) a way to combine packets from multiple voice channels into a
single frame, which produces surprising reductions in the PPS metric.
The Layer 2 FR headers for individual connections collapse into 3-byte
subheaders that replace the other three headers. One L2 header carries
all the channels across the link. (My book "Voice Over Frame Relay" has
- Uncompressed voice packets (G.711) have a payload of 160 bytes.
Quite a few can fit into one MTU of 1500 bytes, saving lots of L2
headers. Many more channels of compressed voice (G.729) with 20-byte
payloads of fit within the MTU.
- Xip-Link, Inc. applies the combination of header compression and
packet aggregation (what they call coalesced packets) in an appliance
intended for real-time optimization of voice traffic where bandwidth or
PPS limits are significant. Benefits increase with the number of
channels. At 50 or 100 G.729 connections, bandwidth drops by more than
half and the PPS level is reduced by more than 98%. Uncompressed voice
doesn't save as much bandwidth but reduces PPS by a factor of 7 to 10.
- Summarizing the technique, Xip-Link applies ROHC (RObust Header
Compression, RFC 4362) to individual VoIP packets. Multiple packets
with 6-byte ROHC headers fill the payload of a packet on which a new IP
header makes the coalesced packet fully routable. The standard IP
header carries the proper marking for priority (DSCP, differentiated
services code point) and doesn't require that intermediate routers be
configured for header compression.
- Your most likely application for coalesced packets is between
internal sites, particularly on satellite circuits where bandwidth and
PPS capacity is more expensive than on terrestrial fiber. Providers of
SIP trunks could ease the burden on edge routers with this technology,
but we may have to wait for that.
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Telephony and the future Voice Network
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