WAP and SIP
This document provides a summary of the ongoing investigation into the potential interworkings of the Session Initiation Protocol (SIP) and the Wireless Application Protocol (WAP). At present, there are no definite conclusions to the investigation; there are still technical questions, mainly in the WAP arena, that need to be answered.
SIP Session Initiation Protocol (SIP) is used to establish sessions between multiple parties in a location-independent manner. The sessions are typically voice sessions, however they could be chat sessions (text based), instant messaging sessions, an online gaming session, etc.
Figure 1
Wireless Application Protocol (WAP) is designed to tackle the limitations of the mobile (cellular) network, in order to allow mobile devices to access internet-based services. Typical services might be email, simple web access, stock alerts, etc.
Figure 2
The WAP protocol is outside of the scope of call signalling and audio transport. Therefore there is no direct mapping between the WAP protocol and SIP. WAP does not contain an equivalent of the SIP INVITE message. For SIP to interact with a WAP enabled mobile phone in call setup and manipulation there has to be a mapping between SIP and the cellular signalling protocols, and that occurs in the gateway in Figure 1.
WAP is predominantly about content delivery to the WAP mobile over the wireless network.
Delivering CLI From SIP Client to a WAP Client
If WAP is about content delivery to mobile clients then perhaps WAP can be used to deliver caller information to a WAP device.
The WAP model allows content to be "pushed" to a mobile client.
Figure 3
In the example shown opposite:
- The SIP Client inititates the call targeted at the WAP mobile by sending an INVITE to sip: +1-212-691-8215@wapit.com.
- The INVITE arrives at the SIP Proxy Server in the WAPit.com domain. The Proxy Server consults with the Application Server Broker and determines that the mobile client associated with the sip URL has registered for the CLI delivery service.
- The CLI Delivery Service is then able to push WAP content to the WAP Push Proxy Gateway asscoiated with the callee. The Push Proxy Gateway then delivers the WAP content over the Wireless Network.
- In parallel to this activity, the Proxy Server, converts the sip URL into a tel URL of the format tel: +1-212-691-8215. The Application Service Broker is consulted again in order to identify a SIP/PSTN gateway that can act as an intermediary between the IP and PSTN networks.
- A Location Server is used to identify a suitable gateway using the TRIP (Telephone Routing over IP) protocol. The WAP client will receive the pushed content containing the callee CLI, and a corresponding incoming call.
- The WAP user is then free to accept or decline the call using the standard phone interface or via WAP content, stored on the mobile client, which is associated with mobile network events, e.g. incoming call.
SIP / CLI Alternatives
Could the caller details (SIP URL) be passed using the CLIP details in the cellular network protocol? If so, this does away with the WAP element for this operation.
Could a SIP client be embedded into the mobile phone, giving the ability to just talk SIP from endpoint to endpoint?
This relies on an IP-based mobile network with high bandwidth. Maybe GPRS will help us with that, but it's more likely that UTMS will be needed. Again this does away with the need for WAP for this application.
SIP > WAP
This scenario shows a SIP client sending an Instant Message to a WAP mobile phone:
- The WAP mobile client subscribes to an Instant Messaging Service provided by WAPit.com. This SIP client sends the MESSAGE to the SIP proxy server with the URL sip: +1-212-691-8215@wapit.com.
- The MESSAGE is sent to the Application Service Broker which is configured to instruct the Proxy Server to forward the MESSAGE onto the Messaging Server/Gateway.
- The Messaging Server/Gateway converts the text associated with the Instant Message into WAP content format and sends it to the WAP Push Proxy Gateway.
- The Push Proxy Gateway then delivers the Instant Message over the wireless network to the WAP client. The messaging gateway is able to interrogate the Push Proxy Gateway to ascertain if the instant message was delivered succesfully.
- Using the result of the interrogation, a SIP response is formulated and returned to the SIP Client via the SIP proxy server.
Figure 4
SIP > WAP Alternatives
Use SMS instead of WAP to push message to mobile client - SMS content is not as rich as WML. For example, a WAP instant message could contain a hyperlink, that could be activated from the WAP client. WAP content could also contain an image.
WAP > SIP
This scenario shows a WAP mobile client sending an Instant Message to a SIP client:
- The WAP client accesses an Instant Message WAP service (WML) via the WAP Gateway.
- The Instant message service is WAP content, which allows a WAP client to enter a message, and specify a destination.
- The WAP client then submits the WML page to the Messaging Server, which extracts the salient information.
- The Messaging server then goes on to form a SIP Instant Message request which is sent via proxy (and ASB) the client acknowledges receipt of message.
Figure 5
SIP / CLI Questions
Can a WAP mobile phone have a WAP session and a telephone call active at the same time?
- It seems that the answer is NO. This is a function of the wireless bearer and the number of transmitters / receivers in the mobile handset. GPRS may alleviate this problem, however I haven't got a definitive answer yet.
Has WAP Push model been implemented in any Push Proxy Gateways?
- +1-212-691-8215@wapit.com.
- The INVITE arrives at the SIP Proxy Server in the WAPit.com domain. The Proxy Server consults with the Application Server Broker and determines that the mobile client associated with the sip URL has registered for the CLI delivery service.
- The CLI Delivery Service is then able to push WAP content to the WAP Push Proxy Gateway asscoiated with the callee. The Push Proxy Gateway then delivers the WAP content over the Wireless Network.
- In parallel to this activity, the Proxy Server, converts the sip URL into a tel URL of the format tel: +1-212-691-8215. The Application Service Broker is consulted again in order to identify a SIP/PSTN gateway that can act as an intermediary between the IP and PSTN networks.
- A Location Server is used to identify a suitable gateway using the TRIP (Telephone Routing over IP) protocol. The WAP client will receive the pushed content containing the callee CLI, and a corresponding incoming call.
- The WAP user is then free to accept or decline the call using the standard phone interface or via WAP content, stored on the mobile client, which is associated with mobile network events, e.g. incoming call.
SIP / CLI Alternatives
Could the caller details (SIP URL) be passed using the CLIP details in the cellular network protocol? If so, this does away with the WAP element for this operation.
Could a SIP client be embedded into the mobile phone, giving the ability to just talk SIP from endpoint to endpoint?
This relies on an IP-based mobile network with high bandwidth. Maybe GPRS will help us with that, but it's more likely that UTMS will be needed. Again this does away with the need for WAP for this application.
SIP > WAP
This scenario shows a SIP client sending an Instant Message to a WAP mobile phone:
- The WAP mobile client subscribes to an Instant Messaging Service provided by WAPit.com. This SIP client sends the MESSAGE to the SIP proxy server with the URL sip: +1-212-691-8215@wapit.com.
- The MESSAGE is sent to the Application Service Broker which is configured to instruct the Proxy Server to forward the MESSAGE onto the Messaging Server/Gateway.
- The Messaging Server/Gateway converts the text associated with the Instant Message into WAP content format and sends it to the WAP Push Proxy Gateway.
- The Push Proxy Gateway then delivers the Instant Message over the wireless network to the WAP client. The messaging gateway is able to interrogate the Push Proxy Gateway to ascertain if the instant message was delivered succesfully.
- Using the result of the interrogation, a SIP response is formulated and returned to the SIP Client via the SIP proxy server.
Figure 4
SIP > WAP Alternatives
Use SMS instead of WAP to push message to mobile client - SMS content is not as rich as WML. For example, a WAP instant message could contain a hyperlink, that could be activated from the WAP client. WAP content could also contain an image.
WAP > SIP
This scenario shows a WAP mobile client sending an Instant Message to a SIP client:
- The WAP client accesses an Instant Message WAP service (WML) via the WAP Gateway.
- The Instant message service is WAP content, which allows a WAP client to enter a message, and specify a destination.
- The WAP client then submits the WML page to the Messaging Server, which extracts the salient information.
- The Messaging server then goes on to form a SIP Instant Message request which is sent via proxy (and ASB) the client acknowledges receipt of message.
Figure 5
SIP / CLI Questions
Can a WAP mobile phone have a WAP session and a telephone call active at the same time?
- It seems that the answer is NO. This is a function of the wireless bearer and the number of transmitters / receivers in the mobile handset. GPRS may alleviate this problem, however I haven't got a definitive answer yet.
Has WAP Push model been implemented in any Push Proxy Gateways?
- What Is SIP Introduction
The Basics
The Session Initiation Protocol (SIP) is a signalling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. The ability to establish these sessions means that a host of innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, Instant Messaging with buddy lists, and IP Centrex services.
Over the last couple of years, the Voice over IP community has adopted SIP as its protocol of choice for signalling. SIP is an RFC standard >>> Overview of SIP in pdf format
Whitepapers
Whitepapers
As and when any SIP-specific white papers become available they will be added to this page. If you have a SIP-related white paper that you would like to add to this page then please contact us .
It is with the help of our sponsors that we continue to develop The SIP Center resource. You will find here our sponsors whitepapers followed by contributions from many other companies and individuals on a wide range of topics that relate to SIP. We hope there is something helpful here for you!
Interested in contributing, send your request to editor@sipcenter.com!
Managing SIP-based applications with WAN link controllers
Communications has always been the foundation for commerce, and today businesses have a variety of communication options. The most powerful form of communications is still face-to-face meetings. However, it is not realistic in terms of cost and logistics, except for the most important interactions. Electronic communication via email and instant messaging are the most convenient and cost-effective, but they lack intimacy.
Multi-site IP Telephony Solutions
A Dual Redundant SIP Service
The Aculab SIP Bridge - For third party call control
Best practices for SIP NAT traversal
AUDIO CODES
"Break Free" - Leveraging SIP in Developing Enhanced Applications
Over the last 15+ years, literally thousands of enhanced applications have been developed for the legacy telecommunications infrastructure. From simple voicemail, to sophisticated contact center solutions, these Computer Telephony Integration (CTI) applications have built value on top of basic PSTN dial-tone, generating substantial revenue in both products and services. However, many of the CTI applications were developed using a restrictive and hard-to-learn architecture that limits the developer’s choices in operating systems, choice of technology suppliers and incurs other serious restraints. This whitepaper outlines a migration strategy that leverages SIP to eliminate many of the past restraints and show how to “break free” from the bonds of the legacy CTI architecture.
Building Applications with SIP - the IP Contact Center
The legacy call center has gone through a metamorphosis, emerging as the IP Contact Center which replaces the PBX and separate IVR and ACD systems and merges email and instant messaging into a new architecture that integrates these functions, leveraging Voice over IP technologies. Whether the goal is to reduce costs in the existing call center, leverage inexpensive overseas labor or add Work At Home Agents (WAHA), IP Contact Centers provide tremendous flexibility to adapt to changing markets and labor resources. By reading this whitepaper, you will learn how SIP can be leveraged as a key enabling technology for the IP Contact Center - delivering scalable and cost effective solutions while avoiding restrictive and expensive API development.
Building Applications with SIP - Conferencing / Collaboration Solutions
Global organizations utilize conference calls as a very important business tool for collaboration. Multi-branch organizations were the first to recognize the value in voice and video conferencing services to economize on travel costs and to coordinate business activities. Other smaller organizations have also begun to recognize that having access to easy-to-use conferencing resources speeds collaboration efforts with clients and suppliers. Whether using a tradition TDM PBX, an IP-PBX or a hosted service provider, SIP is seen as a key technology going forward to help tie organizations together and dramatically reduce the costs of conferencing. By reading this whitepaper you will learn how SIP can be leveraged as a key enabling technology for conferencing and collaboration applications - delivering scalable and cost effective solutions while avoiding restrictive and expensive API development.
Telco VoIP Scalability Test Results... for 10 Million Subscribers
Session Border Control in IMS - An analysis of the requirements for Session Border Control in IMS networks
SIP Market Overview: An Analysis of SIP Technology and the State of the SIP
IP Multicast Explained
VPN Technologies - A Comparison
Managing SIP-based applications with WAN link controllers
Sip Trunking Benefits and Best Practices
A SIP trunk is a service offered by an ITSP to use SIP to set up communications between an enterprise PBX and the ITSP. A trunk includes multiple voice sessions – as many as the enterprise needs. While some see SIP as just voice, SIP trunking can also serve as the starting point for the entire breadth of realtime communications possible with the protocol, including Instant Messaging, presence applications, whiteboarding and application sharing.
(www.ingate.com) Solving the Firewall/NAT Traversal Issue of SIP: Who Should Control Your Security Infrastructure?
Interaction Center Platform® Architecture: Executive Overview
IP Telephony and the Interaction Center Platform®
The Facts About SIP
Multi-Service Domain Protecting Interface Architecture
The Multi-Service Domain Protecting Interface (MSDPI) defines a technology that functions as a secure interface between the network (Cypher Text Domain) and the encryption engine. The technology demonstrated in this report will provide many benefits to the government and private sector.
Session Initiation Protocol Network Encryption Device Plain Text Domain Discovery Service
This report provides a method for cryptographic isolated domains to discover other cryptographic isolated domains by using the IETF Session Initiation Protocol (SIP). This method, called the SIP Network Encryption Device Plain Text Domain Discovery Service (SIP-DS), will not
require a new IETF standard or any modification to existing IETF standards, nor are any specifically configured infrastructure or network devices required.IPsec in VoIP Networks White Paper
SIP Security and the IMS Core
UK Interconnect White Paper
Session Border Control in IMS - based Converged Networks
Solving the Firewall and NAT Traversal Issues for Multimedia over IP Services
IMS SIP and Signaling (Register to download) The RADVISION Perspective - A Technology Overview
Improving Quality in IAD and IP Phone Testing
Benchmarking SIP Server Performance
Henning Schulzrinne, Sankaran Narayanan, Jonathan Lennox, Michael Doyle, Columbia University, Ubiquity.Monitoring and Troubleshooting VoIP Networks with a Network Analyzer
Promiscuous Monitoring in Ethernet and Wi-Fi Networks
IMS Converged Services Gateway
SIP to TCAP Gateway
Multi-site IP Telephony Solutions
A Dual Redundant SIP Service
The Aculab SIP Bridge - For third party call control
Best practices for SIP NAT traversal
AUDIO CODES
"Break Free" - Leveraging SIP in Developing Enhanced Applications
Over the last 15+ years, literally thousands of enhanced applications have been developed for the legacy telecommunications infrastructure. From simple voicemail, to sophisticated contact center solutions, these Computer Telephony Integration (CTI) applications have built value on top of basic PSTN dial-tone, generating substantial revenue in both products and services. However, many of the CTI applications were developed using a restrictive and hard-to-learn architecture that limits the developer’s choices in operating systems, choice of technology suppliers and incurs other serious restraints. This whitepaper outlines a migration strategy that leverages SIP to eliminate many of the past restraints and show how to “break free” from the bonds of the legacy CTI architecture.
Building Applications with SIP - the IP Contact Center
The legacy call center has gone through a metamorphosis, emerging as the IP Contact Center which replaces the PBX and separate IVR and ACD systems and merges email and instant messaging into a new architecture that integrates these functions, leveraging Voice over IP technologies. Whether the goal is to reduce costs in the existing call center, leverage inexpensive overseas labor or add Work At Home Agents (WAHA), IP Contact Centers provide tremendous flexibility to adapt to changing markets and labor resources. By reading this whitepaper, you will learn how SIP can be leveraged as a key enabling technology for the IP Contact Center - delivering scalable and cost effective solutions while avoiding restrictive and expensive API development.
Building Applications with SIP - Conferencing / Collaboration Solutions
Global organizations utilize conference calls as a very important business tool for collaboration. Multi-branch organizations were the first to recognize the value in voice and video conferencing services to economize on travel costs and to coordinate business activities. Other smaller organizations have also begun to recognize that having access to easy-to-use conferencing resources speeds collaboration efforts with clients and suppliers. Whether using a tradition TDM PBX, an IP-PBX or a hosted service provider, SIP is seen as a key technology going forward to help tie organizations together and dramatically reduce the costs of conferencing. By reading this whitepaper you will learn how SIP can be leveraged as a key enabling technology for conferencing and collaboration applications - delivering scalable and cost effective solutions while avoiding restrictive and expensive API development.
Telco VoIP Scalability Test Results... for 10 Million Subscribers
Session Border Control in IMS - An analysis of the requirements for Session Border Control in IMS networks
SIP Market Overview: An Analysis of SIP Technology and the State of the SIP
IP Multicast Explained
VPN Technologies - A Comparison
Managing SIP-based applications with WAN link controllers
Sip Trunking Benefits and Best Practices
A SIP trunk is a service offered by an ITSP to use SIP to set up communications between an enterprise PBX and the ITSP. A trunk includes multiple voice sessions – as many as the enterprise needs. While some see SIP as just voice, SIP trunking can also serve as the starting point for the entire breadth of realtime communications possible with the protocol, including Instant Messaging, presence applications, whiteboarding and application sharing.
(www.ingate.com) Solving the Firewall/NAT Traversal Issue of SIP: Who Should Control Your Security Infrastructure?
Interaction Center Platform® Architecture: Executive Overview
IP Telephony and the Interaction Center Platform®
The Facts About SIP
Multi-Service Domain Protecting Interface Architecture
The Multi-Service Domain Protecting Interface (MSDPI) defines a technology that functions as a secure interface between the network (Cypher Text Domain) and the encryption engine. The technology demonstrated in this report will provide many benefits to the government and private sector.
Session Initiation Protocol Network Encryption Device Plain Text Domain Discovery Service
This report provides a method for cryptographic isolated domains to discover other cryptographic isolated domains by using the IETF Session Initiation Protocol (SIP). This method, called the SIP Network Encryption Device Plain Text Domain Discovery Service (SIP-DS), will not
require a new IETF standard or any modification to existing IETF standards, nor are any specifically configured infrastructure or network devices required.IPsec in VoIP Networks White Paper
SIP Security and the IMS Core
UK Interconnect White Paper
Session Border Control in IMS - based Converged Networks
Solving the Firewall and NAT Traversal Issues for Multimedia over IP Services
IMS SIP and Signaling (Register to download) The RADVISION Perspective - A Technology Overview
Improving Quality in IAD and IP Phone Testing
Benchmarking SIP Server Performance
Henning Schulzrinne, Sankaran Narayanan, Jonathan Lennox, Michael Doyle, Columbia University, Ubiquity.Monitoring and Troubleshooting VoIP Networks with a Network Analyzer
Promiscuous Monitoring in Ethernet and Wi-Fi Networks
IMS Converged Services Gateway
SIP to TCAP Gateway