Abstract:
Systems and methods are disclosed for differentiated handling of VoIP call control messages according to their importance and functionality, thus providing to VoIP infrastructures a level of robust call control similar to that in PSTN networks. SIP messages are classified by their type or content, and resources of a VoIP call control server are allocated to the SIP messages according to a policy. The scheme also provides overload protection and prioritized handling of certain kinds of requests in VoIP call control servers.

Description:
FIELD OF THE INVENTION 
   The present invention is directed to the field of computer networks. It is more particularly directed to Voice-over-Internet Protocol (VoIP) networks using the Session Initiation Protocol (SIP). 
   BACKGROUND OF THE INVENTION 
   Traditional telephony carriers (Public Switched Telephony networks or PSTN) are moving towards a packet-based VoIP infrastructure. A key component of any telephony infrastructure is Call Control. Call control includes call setup and teardown. In VoIP, call control may be achieved through SIP. SIP is a standard protocol for VoIP messages. The SIP messages are text-based. 
   A key requirement for a robust VoIP call control infrastructure is overload control. Since VoIP is an emerging area and different components of the infrastructure are presently in a trial phase, large scale deployments of VoIP has not yet taken place. Once large scale deployments begin, call control in VoIP infrastructure will become critical in avoiding service outages and loss of revenue. Since VoIP is an emerging area, there has been little work on overload protection of VoIP servers. 
   In addition to landline telephony services, SIP has also been chosen as a signaling mechanism for third generation (3G) mobile telephony services, specifically by the 3GPP standards body in its release UMTS R00. The implication of this development is that overload protection for the SIP-based call control infrastructure is needed both for landline and mobile telephony services. A second implication is that in mobile telephony services, the control infrastructure has to support not only call setup and teardown, but the control infrastructure also has to handle terminal mobility between different cell boundaries. This support for hand-off represents an additional load on the call control servers. 
   Another important development is the use of SIP to carry instant messages (IM) and for supporting Presence. Presence is a term understood by persons of ordinary skill in the art and refers to the concept of letting interested users in the system know who is online/offline. For example, a SameTime server shows a “green” light next to someone logged in. Two methods are currently under standardization at the IETF standards body. One method, called SIMPLE, carries Instant Messaging payloads in SIP messages that conceivably will be processed by the same call control infrastructure, as networks moved towards offering converged voice and data services. Using SIP for Instant Messaging and Presence implies an additional load on the SIP based call control infrastructure. 
   In addition to handling call setup, teardown, hand-off and Instant Messaging, SIP call control servers need to also exchange routing information with other call control servers in order to route call setup messages appropriately. 
   For the reasons set forth above, there is a need in the art for a method and system of call control for SIP messages for VoIP. 
   SUMMARY OF THE INVENTION 
   It is thus an object of the present invention to classify SIP messages and place them in separate queues before the messages are processed by a call control server, thereby enabling partitioning of the server resources according to importance and functionality of the messages instead of handling messages in a first-in, first-out arrival sequence. 
   A second object of the present invention is the text-based classification of SIP messages according to the value of individual fields in a SIP message. 
   A third object of this invention is to classify SIP messages according to the intended function of the message. The functions may include terminal registration (SIP REGISTER), call setup (SIP INVITE), call teardown (SIP BYE), terminal mobility (SIP RE-INVITE), and Instant Messaging carried as payload in SIP messages. 
   A fourth object of the present invention is to apply different service rates to the individual queues holding SIP messages according to their intended function. 
   A fifth object of the present invention is to provide the highest priority to emergency calls (911) in a packet-based voice infrastructure. 
   These and other objects, features, and advantages of the present invention will become apparent upon further consideration of the following detailed description of the invention when read in conjunction with the accompanying drawing figures. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  illustrates the prior art handling of SIP messages; and 
       FIG. 2  illustrates the handling of SIP messages in accordance with one embodiment of the present invention. 
   

   DESCRIPTION OF THE INVENTION 
     FIG. 1  illustrates the handling of a SIP message queue  100  according to the prior art. The controller  201  simply responds to each of the SIP control messages  101 - 104 . For example, the SIP message may be a request for a call hand-off in a cellular telephony system. The controller  201  receives the SIP message and accordingly orders the hand-off of the call associated with the message. 
   In the event that the SIP message  104  is associated with a 911 call, the prior art only provides a first-in, first-out handling of the SIP control messages. If the delay in processing the SIP messages  101 - 103  is large enough, the 911 call could be lost due to slow responsiveness to the hand-off request. 
   The present invention allows the differentiated handling of SIP call control messages. A person of ordinary skill in the art would recognize that key SIP messages may include INVITE (session initiation), reINVITE (session modification), REGISTER (bind an user to an address) and BYE (session termination). A combination of reINVITE and REGISTER may be used for handling user/terminal mobility. Incoming requests may be classified and tagged according to a policy specification. These requests can be considered analogous to web requests and can be handled according to a prioritization policy along with isolation of resources to handle each type of message. 
     FIG. 2  illustrates an embodiment of the system and method of the present invention. The invention may be viewed as a soft switch  400  receiving a stream of SIP call control messages  200 . The stream  200  may also be referred to as a SIP server queue. A softswitch is a term understood by persons of ordinary skill in the art and it may include a SIP proxy, call control, user registration, etc. A softswitch essentially subsumes the role of what was conventionally accomplished through a telephony switch. The switch  400  may include a classifier  420 ; a number of token buckets  421 - 423 ; a controller  401 ; and a CPU scheduler. 
   The classifier  420  may be embodied as a software module, microcontroller, or filter that separates incoming messages according to the type of message. Because SIP messages are text-based, the incoming messages can be classified based not only on SIP, but also on specific types of application. The classifier  420  may also operate on other signaling protocols for communications setup defined in the future. 
   The token buckets  421 - 423  may be implemented as queues waiting to be processed by the controller  401  in order of importance or prioritization. The controller  401  reads the call control requests from the incoming messages and may accordingly perform call switching functions. 
   In the illustrated embodiment, given a prioritization of queues, the CPU scheduler  403  implements a controller  401  scheduling policy. The controller  401  represents threads running on the CPU associated with the switch and the CPU scheduler  403  schedules the threads for execution according to a scheduling policy. For example, the scheduler  403  may assign a processing percentage time of 50% processing time to queue  421 , 30% to queue  422 , and 20% to queue  423 . The controller  401  may then process the requests accordingly. Still referring to  FIG. 2 , a description of one embodiment of the methodology of the present invention follows. 
   The architecture of the present invention includes inbound traffic control on SIP server queues to regulate processing of SIP messages, as described above. These messages go through a policer (e.g., the classifier  420 ) that may perform message overload protection by enforcing a leaky bucket flow control onto the incoming message stream. The tagged messages enter the system  400  and compete for processing resources. Service differentiation and QoS guarantees are provided to various classes of workload by scheduling them appropriately. The tags on various messages are visible to these schedulers, allowing them to treat each class of work differently according to its performance requirements. 
   As mentioned above, since SIP messages are text-based, the incoming messages can be classified based not only on SIP message types, but also on specific types of application. For example, 911 calls (sessions) can be accorded the highest priority (e.g., placed in queue  421  with a processing time of 50%). Similarly, SIP INVITE messages that require routing the call through a separate domain (e.g., an MCI caller to a AT&amp;T callee) may be subject to bilateral peering arrangements on the rate of calls that the second domain has pre-negotiated. Thus different policies (rates) may need to be enforced on calls that terminate within the same domain compared to those which require SIP messages to be forwarded to another domain. Another requirement for partitioning of resources (computing/call processing) is driven by handling mobile calls versus regular fixed-line sessions (calls). This would be determined by the token bucket rates of the queues for handling SIP INVITE messages and SIP reINVITE messages. SIP messages that use inter-domain softswitch signaling (SIP BCP-T) are another set of messages that may need to be rate-controlled and guaranteed a minimum share of call-processing resources. 
   An example SIP INVITE message is shown below. The example highlights the text-based nature of the SIP protocol, which allows for service differentiation of SIP messages based on message content as well as message type.
     INVITE sip:bob@biloxi.com SIP/2.0   Via: SIP/2.0/UDP 12.26.17.91:5060   Max-Forwards: 70   To: Bob &lt;sip:bob@biloxi.com&gt;   From: Alice &lt;sip:alice@atlanta.com;tag=1928301774&gt;   Call-ID: a84b4c76e66710@12.26.17.91   CSeq: 314159 INVITE   Contact: &lt;sip:alice@atlanta.com&gt;   Content-Type: application/sdp   Content-Length: 142
 
In this example, this message represents a new call setup request since it is an INVITE message. The classifier  420  will place it in the queue for handling new call setups.
   

   The present invention thus includes a switching method for differentiated services that provides VoIP networks a level of service guarantees so far available only in Public Switched Telephony Networks (PSTN). 
   While the invention has been described in terms of a single embodiment, those skilled in the art will recognize that the invention can be practiced with modification within the spirit and scope of the appended claims. Further, it is noted that, Applicants&#39; intent is to encompass equivalents of all claim elements, even if amended later during prosecution.