Call Continuity from Mobile GSM/LTE network to VoIP/WebRTC

WebRTC is an evolving technology which promises simplified communication platform and stack for developers and hassslefree experience for users. It has the potential to provides in-context, call routed to the best personnel in service calls. Real time mapping of caller’s IP , locations and source metadata can be used for IVR eliminated. Such a complete collaboration tool is possible through WebRTC which is easy set-up, requires no installation no pugins and no download. Extremely secure, WebRTC can interoperate with existing VoIP, video conferencing and even PSTN. The only concern is the Integration with legacy PSTN and teleco environment.

In the present age of IP telephony when telecom convergence is the big thing all around the world, need of the hours is to enable fixed and mobile Service Providers ( SP )  to monetize the subscriber’s phone by extending it to new web based services. SPs can offer a WebRTC Communicator endpoint that uses the same phone number as the subscriber’s fixed or mobile phone. Advanced features enable calls to be transferred between fixed-line, mobile and WebRTC endpoints.

Position of WebRTC on Network protocol stack.

GSM is incompatible with WebRTC media stream due to legacy codecs, even if the WebRTC UA was to support these codecs the signalling translation will be a dffucult feat. Signalling is used for subscriber mobility, subscriber registration, call establishment, etc. Mobile Application Part (MAP), Base Station System Application part (BSSAP), Direct Transfer Application part (DTAP), ISDN User Part (ISUP) are some of the protocols making up GSM. In my opinion Some of the ways to integrate WebRTC to GSM backened could be

  • Develop GSM-To-IP Interworking Component and integrate it with GSM network components (like BTS ).
  • Integrate solution with H.323 based VoIP (Voice Over IP) components like Gatekeeper, Gateways/PBXs, to provide a complete voice/data network solution

Using telco service provider’s SIP trunk , if available, is the easiest way to conect to such backened communication systems.

  • A interface – connection between MSC and BSC;
  • Abis interface – connection between BSC and BTS;
  • D interface – connection between MSC and HLR;
  • Um interface – radio connection between MS and BTS.

GPRS/UMTS Mobile Network can be compatible to WebRTC via Data based communication on GPRS gateway.

LTE Network using Evolves Packet Core can communicate with WebRTC using realtime transcoding and SIP (Session initiation protocol) endpoints conneted to core IMS. AnICE server provides the reflexive IP addresses that the WebRTC implementation needs; the signaling gateway converts the WebRTC webapp’s communication into SIP/IMS signaling and the media relay converts the WebRTC media framing into the telco conformant representation.

Interworking between a WebRTC enabled browser and IMS based Telco Backened : A session is established so that the web app sends an initial INVITE, including an SDP offer for the “outgoing” stream, to the gateway. The signalling gateway will reserve the resources from the media relay in both directions. Consequently, the signalling gateway will send an SDP answer to the initial INVITE and create an SDP offer of its own. This SDP offer is carried in a SIP UPDATE. Once the media between the web app and media relay is set, the session will progress towards IMS and will be handled like any other session. At this point, the media relay has mapped two unidirectional “web app streams” into one bidirectional “IMS stream” and will forward all media between the two. The mapping is done for both audio and video streams, meaning that we are able to support both audio and video calls between WebRTC and Telco clients and conferences.

WebRTC bypasses many limitation of earlier p2p (Peer-to-peer media) streaming frameworks like NATS. It opens avenues for innovative cross-platform use cases such as Healthcare, service technicians on call, Retail and financial communications, phone payments and insurance claims. Other applications such as Unified communication and collaboration are applicable for sales, CRM, remote education etc.

Transfer mobile callto WebRTC session
Transfer mobile callto WebRTC session

SPs can offer 3rd Party WebRTC endpoints to access the user’s phone number and subscription . E.g. enable web applications such as Facebook, Amazon or Netflix to allow their users to make/receive calls or messages directly from the web applications

Revenue Streams :

  • monthly fee for access to WebRTC endpoints and for receiving calls from by 3rd Party WebRTC endpoints
  • One time upgrade fees for Accessing the Web service integration with telecom network like a plan upgrade

Brownie points

  • No software is required to be downloaded on the subscriber’s computer, tablet or mobile phone
  • No desktop support required for the service provider

Plans For Consumer Customers:

  • Subscribers can use the WebRTC endpoints on their computers, tablets or mobile phones as a fixed-line device at home, as a desktop solution when away from home and to avoid international tolls when traveling
  • Subscribers can connect their web services (e.g. Websites , Facebook, Amazon, Netflix) to their fixed or mobile services subscriptions using their SP-provided phone number

Plans For SP Enterprise Customers:

  • Enterprises can deploy a WebRTC endpoint for their employees that provides a single corporate communications endpoint that can be connected to any of the corporation’s UC/PBX and Call Recording systems
  • Employees can use the WebRTC endpoint as their office phone at work, home or when traveling
  • Connects to all leading UC/PBX and Recording platforms simultaneously
  • Enterprises can deploy a single WebRTC endpoint across all their UC/PBX and Recording platforms – current and future
  • Easy for IT departments to deploy – no software is required to be downloaded to employees’ computers, tablets or mobile phones
  • Enables corporate policies and features from the WebRTC endpoint including
  • Displaying the corporate identity
  • Routing calls via corporate networks
  • Tracking and Recording calls and messages

2G to 3G – generation of telecom

Second generation or 2G of telecom emerged a decade after (1990) its predecessor 1G (1980). Although the history of telecom evolution truely beings with internet and further engineered with PSTN, analog voice and switches we shall omit them discussing here as they are truly legacy now.  You can read more about Legacy telecom here-

Legacy Telecom Networks

I use the term legacy telecom system many a times , but have not really described what a legacy system actually is . In my conferences too I am asked to just exactly define a…

We have seen the evolution of teelcom access networks through generations happening pretty quickly recently. While earlier it was a decade that led to the jump between generations, the recent jumps from 3G to 4G to 5G happening fairly quickly.  In this article let us dive into what enhancements went into 2G and its successor 3G, since

Where 2G is referred to as the GSM era , 2.5 G as the GPRS with GSM era. The following two diagram denote the service operators architecture nodes in both these times .

2G / GSM era

As compared to its predecessor 1G which used FDMA ( Frequency Division Multiplexing ) for channelization , 2G used used TDMA and CDMA for dividing the channels .

Note that in pure 2G there was only circuit switched communication services .


2.5G or GPRS era

The advent of 2.5 G, in later part of 1990s, bought packet switching for data access along with existing circuit switching for voice network. While 1G and pure2G relied solely on circuit switching, now 2.5 G used both circuit switched and packet switching. The speed provided by General Packet Radio Service ( GPRS ) was ~= 50 Kbps.

Digital voice was introduced with multiple access technologies like CDMS ( Core Division Multiple access )


2.75G ( EDGE)

EDGE( Enhanced Data Rates for GSM evolution) was deploying on GSM technologies and was also standardised by 3GPP technologies . EDGE delivers higher bit-rates per radio channel, resulting in a threefold increase in capacity and performance compared with an ordinary GSM/GPRS connection with speed upto 1 Mbps.

In terms of transmission techniques, EDGE and its varients used Gaussian minimum-shift keying (GMSK), EDGE uses higher-order PSK/8 phase shift keying (8PSK) for the upper five of its nine modulation and coding schemes.

Note that the processes such as billing etc had begun merging for both the circuit switched and packet switched networks .


Even though 2G evolution was enough to sustain voice abd video calls, the mobile industry became “smarter” and data hungry for faster services ( mobile gaming , video conferencing ,video streaming, social media interactions are some of the usecases ). It became necessary to bring in faster speed while evolving towards and hence was born 3G in early 20000. Some of the tecehnolgies which were branded 3G are

 UMTS (Universal Mobile Telecommunications System) 

Core technology for 3G ,


3.5G ( HSPA)

Now 3G was further succeeded by 3.5G ( HSPA – High Speed Downlink Packet Access ) with max theoritical 21.6 Mbps.

Eventually 4G ( LTE Long Term Evolution ) overtook the indutry with newer technologies but the impressive array of technologies in transaition between 2G to 3G to 4G was awe inspirinig indeed .

References :

Also Read

4G/Long Term Evolution (LTE), VOLTE

LTE stands for Long Term Evolution and is a registered trademark owned by ETSI (European Telecommunications Standards Institute) for the wireless data communications technology and a development of the GSM/UMTS standards.

5G and IMS

striking features of 5G – entirely IP based ability to connect 100x more devices ( IOT favourable ) speed upto 10 Gbit/s high peak bit rate high data volume per unit area virtually 0 latency hence high response time