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Transporting Encapsulated Voice Traffic November 29, 1999 By Chris Lewis It's hard to remember a time when there wasn't some hoopla surrounding voice and data integration; just how long has it been now that we've been hearing predictions that voice and data networks will merge "within five years?" Finally, real options exist to combine voice and data on one network. But don't even think about cutting the cord to your PBXes yet. The best approach is to evolve toward integration rather than make wholesale changes. H.323-standard Ethernet telephones, with supporting IP network-aware devices, are destined to make the PBX obsolete. However, it will take many years to get this setup to match the reliability and ease of use of PBX facilities, including conveniences such as call forward on busy or no answer, voice mail and ACD (Automatic Call Distribution).
A first stage in integrating the voice and data traffic may be to transport voice between corporate headquarters and the branches over the existing data network. The goal is to eliminate charges for interoffice or interbranch calls. Transporting voice traffic over existing data links raises a few issues pertinent to any installation of this type. We'll use the Cisco Systems MC3810 concentrator as an example of how one vendor addresses these issues. We have implemented packetized voice solutions with this device and so are more familiar with it than with competing products. In our example, the Cisco MC3810 performs two functions: It acts as the site router for connecting branches to the corporate office and it provides emulated tie-line connectivity between the PBXes. In the configuration shown below, IP data traffic is routed between the WAN and LAN interfaces. The voice traffic, however, is not encapsulated within IP datagrams. Instead, it is encapsulated within the Layer 2 frame type in use and, therefore, can only be transported on a point-to-point link, as would any other tie-line.
Physical Connectivity
To check that physical connectivity is OK, we want to see that a voice port connected to the PBX tie-line port shows a seized condition when the PBX tries to use the MC3810 as a tie line. Seized, which is another voice-communications term, means that the port is in use. Assuming the physical connections are correct and the PBX has been configured for tie-line operation, we can test the connection as follows: A handset on the PBX dials the tie-line number associated in the PBX configuration with the MC3810 connection (for argument's sake, let's make this 83) and is then presented with a dial tone generated by the MC3810. If this happens, the MC3810 voice port has been seized. There should be a display on the device you are using to show this change of status. The voice-port summary command on the 3810 is illustrated below.
If this does not work and the PBX and physical connections are correct, you must delve into the voice-port configuration. Thankfully, it is not too complicated, as there are only a few variables with which to experiment. The settings to look at on the voice port are two-wire or four-wire operation (the cabling illustrated in "Conductor Connections" assumes four-wire operation), E&M signal type and signal mode. The MC3810's default settings work for the majority of PBXes. However, the default two-wire operation needs to be set to four-wire operation: This voice-port configuration will work with most PBXes in the United States. But if you are deploying this type of solution internationally, more adjustments are necessary. Domestically, PBXes use E&M Type 1 signaling. PBXes in the United Kingdom, however, use E&M Type 5 (known as DC-5 there). The other setting that may need to change is the signal mode. The default is "wink-start," which you may need to change to "immediate" for some PBXes. The key is to marry the voice-port configuration options between the MC3810 (or equivalent) and the PBX being used. The poor quality of most documentation in this area means you must often resort to trial and error.
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In "MC3810-to-PBX Connectivity" (above right), you see a corporate headquarters with branches in the United States as well as abroad. Each branch requires access to central database applications, which justifies the cost of maintaining some type of network to the branches. It does not matter whether this access is over leased lines, frame relay or ATM. Each branch has its own smaller PBX to service the users' needs at that location.
This will seem curious to data-only engineers. The terms E&M, tip and ring do not appear in data networks. But in practice there is little to know. I have heard E&M referred to as ear and mouth, to indicate the purpose of the leads on a voice system. However, some think it originated from rEceive and transMit. Tip and ring are voice-only terms that refer to signaling conductors.









