VL-83

Efficient SIP Trunks

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:

Bandwidth

Consider the relative size of the payload (voice information) and the packet headers (routing and processing information).

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 private network.

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 second (Mpps).

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.

Compress/Combine

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 the details.)

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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