Although the history of telecom evolution begins with PSTN and switches we shall oit them as they are truly legacy now . We have seen the evolution of second to third generation of telecom most recently . Where 2 G 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 .
Note that in pure 2G there was only circuit switched communication services .
The advent 2.5 G bought packet switching for data access along with existing circuit switching for voice network .
Note that the processes such as billing etc had begun merging for both the circuit switched and packet switched networks .
However as the mobile became smarted and hungry for faster internet , it bbecame necessary to bring in faster speed and hence was born 3G. . Now 3G was further succeeded by 3.5G ( HSPA – High Speed Downlink Packet Access ) eventually 4G ( LTE Long Term Evolution ) as we can see now but that is another story .
A lot of service providers ie telecom operators had deduced their own ways to provide Web based communication even before WebRTC was born . With time , as WebRTC has become stronger , more secure , resilient to failure they have come around to migrate their existing system from previous closed box native APIs to opensource WebRTC APIs.
The first figure ( given below ) depicts a communication platform build over plugins and proprietary APIs using HTTP REST based signaling .
As the migration took place the proprietary API components were replaced by Open standard based entities such as plugins were replaced by WebRTC APIs, HTTP REST based signalling was replaced by SIP ( Session Initiation Protocol ) .
Note telecom operator network did not had to face transformation by integration of WebRTC elements .
SIP Server convert the SIP transport from WebSocket protocol to UDP, TCP or TLS which are supported by all legacy networks. It also facilitates the use of rich serves such as phonebook synchronisation , file sharing , oauth in client .
How does WebRTC Solution traverse through FireWalls ?
NAT traversal across Firewalls is achieved via TURN/STUN through ICE candidates gathering .Current ice_servers are : stun:stun.l.google.com:19302 and turn:firstname.lastname@example.org
What audio and video codecs are supported by WebRTC client side alone ?
Without the role of Media Server WebRTC solution supports Opus , PCMA , PCMU for audio and VP8 for video call.
RTCBreaker if enabled provides a third party B2BUA agent that performs certain level of codec conversion to H.264, H.263, Theora or MP4V-ES for non WebRTC supported agents.
What video resolution is supported by WebRTC solution ?
The browser will try to find the best video size between max and min based on the camera capabilities.
We can also predefine the video size such as minWidth, minHeight, maxWidth, maxHeight.
What bandwidth is required to run WebRTC solution ?
We can set maximum audio and video bandwidth to use or use the browser’s ability to set it hy default at runtime . This will change the outgoing SDP to include a “b:AS=” attribute. Browser negotiates the right value using RTCP-REMB and congestion control.
List of Web based SIP clients
SIPML5 client by Dubango
Telestax WebRTC client
SIPJS with flash network support
SIP phones in Ubuntu / Linux
Yate SIP phone
There are ready made build of Linphone for Windows , Mac and Mobile
Aletrnatively one can also build the Linphone from source
[ 57%] Performing configure step for 'EP_ms2'
loading initial cache file /home/altanai/linphone-desktop/WORK/WORK/desktop//tmp/EP_ms2/EP_ms2-cache-RelWithDebInfo.cmake
CMake Error at CMakeLists.txt:322 (message):
Could not find a support sound driver API. Use -DENABLE_SOUND=NO if you
don't care about having sound.