SIP Tips – How SIP Works Tutorial

May 20, 2008
Since too few people really know how SIP works, here is the text explanation of each step of the way. After the IP phone is plugged into the LAN network and powers up via the electrical outlet or via POE-Power Over Ethernet cable, the phone needs to get an IP-Internet Protocol Address from the DHCP-Dynamic Host Configuration Protocol server.

SIP Signaling
User dials number 303-594-1694. SIP-URI-Uniform Resource Identifier is retrieved from DNS-Domain Name System: sip:tom@xyz.com;Transport=UDP. 
The SIP INVITE along with the SDP-Session Description Protocol is formatted into an Internet Message Format and encapsulated into ethernet and sent via the LAN Switch to the router and encapsulated into IP or IP-MPLS and UDP and the SIP INVITE is sent via TCP, UDP and other protocols to the destination proxy. The caller receives a 100 (Trying) response indicates that the INVITE has been received and that the proxy is working "on behalf of" to route the INVITE to the destination. The caller receives a 180 (Ringing) and begins ear ringing using an audio ringback tone or by displaying a message on the telephone screen.
When the person called picks up the handset, the SIP phone sends a 200 (OK) response to indicate that the call has been answered. The 200 (OK) message contains a SDP-Session Description Protocol media description of the type of session that the other party is willing to establish. An ACK message is sent.
SIP Media Session
User begins talking. Voice is transcoded into a CODEC, e.g. G.711, G.729 based on SDP. Voice is packetized. RTP is added and encapsulated in Ethernet and sent via the LAN Switch to the router and encapsulated into IP or IP-MPLS and UDP. If on-net IP network, the digital data is converted to an optical data stream and sent via optical fiber to an internet or internal router. If off-net, the digital data is channelized using a MG-Media Gateway to a TDM-Time Division Multiplexed channel and sent via an optical data stream to a Class 5 CO-Central Office switch for connections to the PSTN-Public Switched Telephone Network. User hangs up and a BYE message is sent. A 200 ACK is received confirming disconnect.
 
If you want to know more, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online.  The online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available).   For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com.  Discounts are also available to members of the SIP Forum and MS Partners for $99 per student during May.
Courses are free to channel partners – see terms and conditions at http://www.techtionary.com/techu/


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