TSM 350G – IP Telephony Lab
Spring Semester 2013
Objectives
1 – Overall objective: make a call between two VoIP endpoints (or two user-agents in SIP parlance.), capture the wireshark trace, and draw the ladder diagram. The calls can be:
a. A peer-to-peer call (e.g., the call does not go through a proxy of any sort, the SIP and RTP go endpoint to endpoint.)
b. A call between two endpoints served by the same SIP proxy/location server (for example, two IP Phones served by an IP-PBX). In this case the call might be made using an abbreviated dial plan for between two phones on the same subnet. (If you want to do this type of call, the TAs will have an IP-PBX set up in the lab for you to use).
c. A call between two endpoints that are registered to different location servers/proxies. (For example, between a soft-client registered to Google Talk, and another registered to an IP-PBX that has a trunk to Google.)
d. Some other permutation of (a-c).
2 – To accomplish the overall objective, in addition to picking a model (a-d), you will need to:
· Configure two endpoints (your choice of VoIP phones, ATAs, softclients, etc.), and have them register to the correct location server (assuming you are using a loc server)
· Figure out how to get the wireshark traces, and how to use wireshark (one way is to make sure that each endpoint is plugged into a hub). Wireshark can be downloaded from http://www.wireshark.org/download.html, make sure that you also have the appropriate pcap file for your O/S. To capture traces on a WiFi connection, you will need a WiFi dongle that allows you to echo the traffic – we have one that you can borrow on pain of instant and painful death if it’s not returned. Also, I hear there are problems with the pcap file if you are running Windows 8.
· As an alternative to wireshark, you are also welcome to use ngrep.
3 – What are you trying to learn:
· How to use a protocol analyzer
· How to configure a SIP endpoint
· How to read SIP traces, how to build a ladder diagram based on the protocol messages
· How to stand on your own feet and figure out what’s going on in an IP environment/network
You are welcome to do work in the lab, or do it in the comfort of your own home. If you chose to do it at home, you may borrow VoIP phones, ATAs, and hubs by checking them out with the lab TAs. (Lab TAs – the phones/clients are in the locked cabinet, please make sure that you have a written record of checking things out and then in again).
If you decide to use a softclient for one of your endpoints (a good idea), there are lots of them out there. The original counterpath (XTEN) client can be found here (it’s no longer posted by Counterpath, but lots of folks like it because it provides SIP debugs directly in the client. Another interesting choice is to try an VoIP client for your smartphone, for example http://sipdroid.org/ (although there are many).
3 – Make sure that you save your wireshark traces and ladder diagrams. There will probably be a quiz on this lab - which will require you to submit both the traces and your ladder diagram, as well as answer a few questions. You are welcome to ask questions and get help from members of the class, the lab TAs, or anyone else you know – but each student needs to have their own setup, collect their own data, and be able to share the data and speak (or write) to it clear.
Figure 1 – here’s an example of a peer-to-peer call between an ip phone and an ATA.
Figure 2 – calls from VoIP phone to VoIP phone using an IP-PBX. Note that the SIP and RTP flows are meant to be illustrative (the IP-PBX acts as a B:B UA for signaling, but media goes end to end). Obviously all the packets flow through the hub and the router.
Phone Configurations
CISCO PHONEs: To change the sip proxy address on the cisco phone, press the settings key, scroll to the bottom of the menu, and press the unlock key to allow you to change the phone settings from the front of the phone. The phone unlock password is 1234. Then scroll up to SIP Configuration => Line 1 Settings => Proxy Address, and change the proxy address so that it points to the desired address. You can also edit the Cisco configuration by editing the xml config file that gets pulled down to the phone. In this case you need to add the URL of the config file server via the phone menu .
CISCO ATAs: The ATA (like many IP phones and Internet Access Devices) is configured using a web server running on the ATA. Point a web browser at http://ip_address_of_ata/dev, and change the GkOrProxy address to be the desired address to send calls to.
Cisco ATA configuration
Polycom Phones: Polycom phones can be configured either by using the web browser on each phone, or by editing the XML in the /tftpboot directory. You can You can hand-edit the Polycom configs, but I would recommend the use of an xml editor. I would suggest you try both.
Polycom Phone Web Browser
Counterpath XTEN client .. has built in window for SIP messages
Use of an XML editor for editing Polycom configs