SIP Session Initiation Protocol – II

How Does it Work?
SIP provides connectivity and access. It enables communications between two SIP devices on a peer-topeer basis, and acts as a client server by allowing SIP end users, called user agents (UA) to act as clients when initiating a request, or as servers when responding to a request. This could mean an attempt to call a SIP device results in a redirection to another device. The SIP infrastructure can have SIP servers to act as proxies, gateways, registrars and redirectors to bridge into multiple devices or other environments such as linking to the public switched telephone network (PSTN) or a corporate PBX. On a peer-to-peer basis, SIP makes an invitation, which is either acknowledged or cancelled. Once the set up is complete, the Real Time Protocol (RTP) takes over to provide the actual communications. When the call is completed, the user hangs up and the SIP “Bye” message is sent out to end the session. A diagram of a simple SIP peer-to-peer call follows.

Session Initiation Protocol – SIP



Calling into a non-SIP environment requires registering to be accessible on more than one device, or for changed locations. In this case, a SIP proxy server provides services such as redirecting a call to where you have moved, launching parallel requests (forking) to more than one device or connecting into a non-SIP device such as an analog or digital PBX telephone (SIP gateway). SIP proxy servers also track users by dynamically updating changes and noting when the user is on line. This is the concept of “presence” and it enables a user to be found regardless of physical location. The primary function of SIP is to set up a connection. How communications take place depends on the device being used, the user’s status, and location in the network (presence) and the preferences listed when the user registered. For example, the manager might be on a conference call at his/her desk. If the second line rings and the manager puts the conference call on hold to speak with a client, the other participants have to tolerate music on hold until the conference operator can mute the manager’s port. Instead of disrupting a conference call the manger’s profile would state that if a call comes in from this particular client, identified by using caller ID, the caller has the option of engaging the manager in a chat session. An instant message on the manager’s PC screen might appear saying a client is trying to reach the manager (an “invite”) and he/she could accept (an ACK). The client could receive a message that says the manager is available to chat and see on his/her PC that a chat session has been set up. All of this depends on the manager’s network, devices, and desktop capabilities.
“There is already rapid growth in the number of novel and useful functions in new SIP-based systems,” said Jay Batson, founder and Chairman of Pingtel Corp., a pioneer and leader in SIP phones and technology. “Emerging SIP products are highly flexible and customizable, and are also rigorously tested for multi-vendor interoperability. This combination enables customers to find and deploy products that enhance their businesses much more simply and easily than with traditional voice systems.” SIP is an evolving standard that is slowly gaining in acceptance and use in the telecommunications industry. The number of popular PBX features it supports is also growing. The cost of SIP phones, although currently higher than digital or IP phones, will decrease as demand increases and soft phones offer more cost-effective ways to realize the benefits of SIP in locations where their use is practical (e.g., call centers). Only recently have SIP telephones begun to reach price points similar to today’s digital PBX phones.

Security, as with any IP-based device or service, is an issue, although there are security measures that can be taken such as embedding a firewall and NAT within the SIP proxy (Windows XP does this) or using stand-alone gateways. Telecommunications service providers must still address the issue of mapping PSTN numbers and SIP addresses. Recently, the Department of Commerce said it would support the electronic numbering system, known as ENUM to allow consumers to specify a single identification system for their telephone numbers, e-mail, fax numbers, cell phone numbers and instant messaging addresses. The ENUM standard, known as E.164.arpa translates telephone numbers to Internet addresses and vice versa. Enhanced 911 services are also a challenge with not just SIP, but all IP telephony devices due to the inherent mobility of IP. Since these devices essentially float in the network, ensuring E911 location information becomes a challenge. Within the SIP community several proposals have been offered ranging from installing GPS chip sets in the devices to having users simply log on to specify their location. Because a growing number of states require location information for E911 calls, this issue calls for some type of standardization across the SIP product line.