Tag Archives: WebRTC

WebRTC CPaaS ( Communication Platform as a Service )

Posted on July 4, 2019 | Leave a comment

A CPasS ( communication platform as a service ) is cloud based communication platform that provides real time communication capabilities. This should be easily integrable with any given external environment or application of the customer, without him worrying about building backend infrastructure or interfaces .

Traditionally , with IP protected protocols , licensed codecs maintaining a signalling protocol stack , network interfaces building communication platform was a costly affair. Cisco , Facetime , Skype were the only OTT ( over the top) players taking away from telco’s call revenue .

However with the advent of standardize , open source protocol and codecs plenty of CPaaS providers have crowded the market making more supply than there is demand. A customer wanting to quickly integrate real time communications on his platform has many options to choose from. This article provides an insight to how CPaaS solution are architectured and programmed

Sample CPass Architecture build on open source technologies

I have written an article before on Steps for building and deploying WebRTC solution , which includes standalone , cloud hosted and TURN based NAT handler systems .

A typical CPaaS solution provides

  • Call server + Media Server that can be interacted with via UA
  • Comm clients like sipphones , webrtc client , SDK ( software development kits ) or libraries for desktop , embedded and/or mobile platforms .
  • APIs that can trigger automated calls and perform preprogrammed routing.
  • Rich documentation and samples to build various apps such as call centre solutions , interactive auto-attendant using IVR , DTMF , conference solutions etc .
  • Some CPaaS providers also add features like transcribing ,transcoding , recording , playback etc to provide edge over other CPaaS providers

Advantages of using a CPaaS vs building your own RTC platform

Tech insights and experiences

companies who have been catering to telco and communication domain make robust solutions based on industry best practices which beats novice solution build in a fortnight anyday

keeping up with emerging trends

Market trends like new codecs , rich communication services , multi tenancy, contextual communication , NLP , other ML based enhancements are provided by CPaaS company

Auto Scaling , High Availability

A firm specializing in CPaaS solution has already thought of clustering and autoscaling to meet peak traffic requirements and backup/replication on standby servers to activate incase of failure

CAPEX and OPEX

using a Cpaas saves on human resources, infrastructure, and time to market. It saves tremendously on underlying IT infrastructure and many a times provides flexible pricing models

In a nutshell I have come across so many small size startups trying to build CPaaS solution from scratch but only realising it after weeks of trying to build a MVP that they are stuck with firewall, NAT, media quality or interoperability issues . Since there are so many solution already out in the market it is best to instead use them as underlying layer and build applications services using it such as callcentre or CRM services etc .

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Posted in WebRTC SaaS

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Kamailio WebRTC SIP Server

Posted on September 4, 2018 | Leave a comment

The purpose of this article if to demo the process of using Kamailio + RTP Engine to enable SIP based WebRTC call to a traditional SIP UA like Xlite. Kamailio Will thus provide not only call routing but also NATing , TLS and websocket support for webrtc endpoints.

For this bridging of SRTP from WebRTC endpoint like JSSIP to RTP for SIP UA like Xlite , we will use RTP engine .

Kamailio

Kamailio with modules like websocket , TLS , NATHelper which help it to support websocket based SIP which default kamailio configuration doesn’t.

Snippets from kamailio config to support webrtc endpoints is below . the detailed explanation is at kamailio config for webrtc 

loadmodule "tm.so"
 loadmodule "sl.so"
 loadmodule "rr.so"
 loadmodule "pv.so"
 loadmodule "maxfwd.so"
 loadmodule "usrloc.so"
 loadmodule "textops.so"
 loadmodule "siputils.so"
 loadmodule "xlog.so"
 loadmodule "sanity.so"
 loadmodule "ctl.so"
 loadmodule "kex.so"
 loadmodule "corex.so"
 loadmodule "tls.so"
 loadmodule "xhttp.so"
 loadmodule "websocket.so"
 loadmodule "nathelper.so"

Configuration of some important modules

TLS module

To provide the SSL support which webrtc endpoints require Certificate Authority CA and provate certs signed by it

creating file paths 

mkdir certs
mkdir certs/private
mkdir certs/newcerts
touch certs/index.txt
echo 01 >certs/serial
echo 01 >certs/crlnumber

list the files 

/home/ubuntu/certs# ls
crlnumber  index.txt  newcerts  private  serial

create ca private key

openssl genrsa -out certs/private/cakey.pem 2048
chmod 600 certs/private/cakey.pem

create ca self signed certificate

openssl req -out certs/cacert.pem -x509 -new -key certs/private/cakey.pem

create server / client certificate, a private key (by name privkey.pem)

openssl req -out kamailio1_cert_req.pem -new -nodes
openssl ca -in kamailio1_cert_req.pem -out kamailio1_cert.pem

output should be like
…
Certificate is to be certified until Jun 25 11:02:41 2020 GMT (365 days)
Sign the certificate? [y/n]:y
1 out of 1 certificate requests certified, commit? [y/n]y
Write out database with 1 new entries
Data Base Updated

and files genrated shoudl look like

/home/ubuntu# ls 
 certs  kamailio1_cert.pem kamailio1_cert_req.pem privkey.pem

copy the newly created certs to their respective paths

mkdir /etc/pki/CA/
cp kamailio1_cert.pem /etc/pki/CA/
cp privkey.pem /etc/pki/CA/

make list of ca certs by finidng all cacerts accross root firectory and appending them to a catlist pem

find / -name cacert.pem
cat /usr/share/doc/libssl-doc/demos/cms/cacert.pem >> /home/ubuntu/catlist.pem
cat /usr/share/doc/libssl-doc/demos/smime/cacert.pem >> /home/ubuntu/catlist.pem
cat /home/ubuntu/kamailio_source_code/misc/tls-ca/rootCA/cacert.pem >> /home/ubuntu/catlist.pem
...
cp /home/ubuntu/catlist.pem /etc/pki/CA/

update kamailio.cfg 

#!ifdef WITH_TLS
enable_tls=1
#!endif
...
modparam("tls", "tls_method", "SSLv23")
modparam("tls", "certificate", "/etc/pki/CA/kamailio1_cert.pem")
modparam("tls", "private_key", "/etc/pki/CA/privkey.pem")
modparam("tls", "ca_list", "/etc/pki/CA/calist.pem")

Websocket module

Websocket is considered a transport option just as TCP or UDP in kamailio config , hence just as one defines IP addr and ports for TCP, UDP protocol , we need to define the same for WS or WSS 

#!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g"
#!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g"
...
listen=MY_IP_ADDR
#!ifdef WITH_WEBSOCKETS
listen=MY_WS_ADDR
#!ifdef WITH_TLS
listen=MY_WSS_ADDR
#!endif
#!endif

check if port in R-URI meant for ws or wss, did not receive websocket or secure websocket 

if (($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) {
    xlog("L_WARN", "SIP request received on $Rp\n");
    sl_send_reply("403", "Forbidden");
    exit;
}

request_route for websocket , included checking is client is behind NAT using nat_uac_test methods from NAThelper. If it is then for REGISTER methods do fix_nated_register and for other add_contact_alias

#!ifdef WITH_WEBSOCKETS
if (nat_uac_test(64)) {
	# NAT traversal  WebSocket
	force_rport();
	if (is_method("REGISTER")) {
		fix_nated_register();
	} else {
		if (!add_contact_alias()) {
		xlog("L_ERR", "Error aliasing contact <$ct&gt;\n");
		sl_send_reply("400", "Bad Request");
		exit;
		}
	}
}
#!endif

RTP engine

RTP relay and NAT helps with RTP packets

For detailed steps goto https://telecom.altanai.com/2018/04/03/rtp-engine-on-kamailio-sip-server/

dependencies

apt-get remove rtpproxy
sudo apt install debhelper iptables-dev libcurl4-openssl-dev libglib2.0-dev libxmlrpc-core-c3-dev libhiredis-dev markdown build-essential:native

source 

git clone https://github.com/sipwise/rtpengine.git
cd rtpengine
 ./debian/flavors/no_ngcp

run

rtpengine --interface=54.86.35.95 --listen-ng=25061 --listen-cli=25062 --foreground --log-stderr --listen-udp=2222 --listen-tcp=25060

integrate with kamailio using 

loadmodule "rtpengine.so"
...
modparam("rtpengine", "rtpengine_sock", "udp:localhost:7722")
modparam("nathelper", "received_avp", "$avp(s:rcv)")
...
tbd

JSSIP

Like most other WebRTC libraries , JSSIP is event driven and provides provide core WEBRTC API like getUserMedia and RTP PeerConnection providing STUN,ICE,DTLS, SRTP features. It also integrated with rtcninja to provide cross browser accessibility. The differentiators with JSSIP lies in the fact that it supports SIP stack over websockets

Building JSSIP client for kamailio

Can use CDN based https://cdnjs.cloudflare.com/ajax/libs/jssip/3.1.2/jssip.min.js

or can take a pull from JSSIP repo and build urself using gulp https://github.com/versatica/JsSIP

instantiate JSSIP websocket interface with kamailio IP

var socket = new JsSIP.WebSocketInterface('wss://<kamailio_ip>:443');

Add configuration for registeration . Note if not using kamailio as proxy to SBC, it is recommended to add regiseteration features to provide user reachability for incoming calls and NAT pings

var configuration = {
sockets : socket,
uri : 'sip:username@example.com',
password : 'password'
};

create UA and start

var ua = new JsSIP.UA(configuration);
ua.start();

SIP over WEBSOCKET messages and kamailio processing

REGISTER sip JSSIP UA with user altanai , domina voiptelcom.com

REGISTER sip:voiptelco.com SIP/2.0
Via: SIP/2.0/WSS 830p2l39g8bg.invalid;branch=z9hG4bK242397
Max-Forwards: 69
To: 
From: ;tag=3jaad0q8l8
Call-ID: fvn2cd1b9gqh6kd7nqdpj5
CSeq: 1 REGISTER
Contact: ;+sip.ice;reg-id=1;+sip.instance="";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 3.1.2
Content-Length: 0

Processed by kamailio using REGISTRAR route block

route(REGISTRAR);
..
route[REGISTRAR] {
	if (is_method("REGISTER")) {
		if (!save("location")) {
			sl_reply_error();
		}
		exit;
	}
}

SIP/2.0 200 OK
Via: SIP/2.0/WSS 830p2l39g8bg.invalid;branch=z9hG4bK242397;rport=19035;received=x.x.x.x
To: ;tag=4dad943d40a0a309c33d64467664aa30.f6d3
From: ;tag=3jaad0q8l8
Call-ID: fvn2cd1b9gqh6kd7nqdpj5
CSeq: 1 REGISTER
Contact: ;expires=600;received="sip:x.x.x.x:19035;transport=ws";pub-gruu="sip:altanai@voiptelco.com;gr=urn:uuid:80fa65e7-1cd7-4e40-bbee-c07f7a1ae9a5";temp-gruu="sip:uloc-5d260578-4b58-c-eeac6bd6@voiptelco.com;gr";+sip.instance="";reg-id=1
Server: kamailio (5.2.3 (x86_64/linux))
Content-Length: 0

INVITE from user1 altanai to john , notice that TO header doesnt have tag . this will handy for recognizing whether it is first message of dialog of=r and indialog message such as ACK , RE-INVITE , BYE etc

INVITE sip:john@voiptelco.com SIP/2.0
Via: SIP/2.0/WSS ipoct61ao12v.invalid;branch=z9hG4bK4220209
Max-Forwards: 69
To: sip:john@voiptelco.com
From: sip:altanai@voiptelco.com;tag=2q0lecmbsn
Call-ID: s8bnv5869fp68d1ju8c1
CSeq: 1799 INVITE
Contact: 
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: timer,gruu,ice,replaces,outbound
User-Agent: JsSIP 3.1.2
Content-Length: 1823

with SDP containing codecs and ICE details. Supporting audio over UDP/ TLS/RTL /SAVPF . Codecs beings

  • 111 OPUS
  • 103 ISAC/16000
  • 104 ISAC/32000
  • 9 G722
  • 0 PCMU / G.711u narrowband
  • 8 PCMA / G.711
  • 106 , 105 , 13 – CN / comfort noise
  • 110 , 112 , 113 , 126 – telephone-event / DTMF
v=0
o=- 4779000713447952953 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS SFIXFrpsOUskJ5JQhp1mIARlDk6S3hVFTOBb
m=audio 55839 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 192.168.0.3
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3802297132 1 udp 2122260223 192.168.0.3 55839 typ host generation 0 network-id 1 network-cost 10
a=candidate:2887880668 1 tcp 1518280447 192.168.0.3 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:0PIq
a=ice-pwd:i7ccvGPXLDO5JqMwbCUqMcyN
a=ice-options:trickle
a=fingerprint:sha-256 AB:50:70:E3:57:E3:0C:7B:61:3B:03:5B:0F:54:14:14:9C:49:50:16:07:DC:E7:09:3E:4D:B5:A0:2B:EC:84:A1
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:SFIXFrpsOUskJ5JQhp1mIARlDk6S3hVFTOBb e803836d-249a-4b81-b73f-17e0f08dde5a
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2800821831 cname:kLWViBLDLVfCXY8x
a=ssrc:2800821831 msid:SFIXFrpsOUskJ5JQhp1mIARlDk6S3hVFTOBb e803836d-249a-4b81-b73f-17e0f08dde5a
a=ssrc:2800821831 mslabel:SFIXFrpsOUskJ5JQhp1mIARlDk6S3hVFTOBb
a=ssrc:2800821831 label:e803836d-249a-4b81-b73f-17e0f08dde5a

100 trying from callee . Note the to and from headers remain same for request or responses . This is send automatically by kamailio for INVITE.

SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/WSS ipoct61ao12v.invalid;branch=z9hG4bK4220209;rport=17502;received=x.x.x.x
To: sip:john@voiptelco.com
From: sip:altanai@voiptelco.com;tag=2q0lecmbsn
Call-ID: s8bnv5869fp68d1ju8c1
CSeq: 1799 INVITE
Server: kamailio (5.2.3 (x86_64/linux))
Content-Length: 0

180 ringing from Callee , note the addition of contact header

SIP/2.0 180 Ringing
Record-Route: 
Via: SIP/2.0/WSS ipoct61ao12v.invalid;rport=17502;received=x.x.x.x;branch=z9hG4bK4220209
To: sip:john@voiptelco.com;tag=pvm73e3t89
From: sip:altanai@voiptelco.com;tag=2q0lecmbsn
Call-ID: s8bnv5869fp68d1ju8c1
CSeq: 1799 INVITE
Contact: sip:john@voiptelco.com;alias=x.x.x.x~17510~6;gr=urn:uuid:2e560b36-3ea8-41fd-80e3-ede66babb8a7
Supported: timer,gruu,ice,replaces,outbound
Content-Length: 0

200 OK with SDP

SIP/2.0 200 OK
Record-Route: 
Via: SIP/2.0/WSS ipoct61ao12v.invalid;rport=17502;received=x.x.x.x;branch=z9hG4bK4220209
To: sip:altanai@voiptelco.com;tag=pvm73e3t89
From: sip:john@voiptelco.com ;tag=2q0lecmbsn
Call-ID: s8bnv5869fp68d1ju8c1
CSeq: 1799 INVITE
Contact: sip:john@voiptelco.com;alias=x.x.x.x~17510~6;gr=urn:uuid:2e560b36-3ea8-41fd-80e3-ede66babb8a7
Session-Expires: 90;refresher=uas
Supported: timer,gruu,ice,replaces,outbound
Content-Type: application/sdp
Content-Length: 1477
v=0
o=- 4562215268128860297 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS oxU77z9z9RfNL4CayvM1cMJKI0r7u6ZdqLBd
m=audio 55380 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 192.168.0.3
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3802297132 1 udp 2122260223 192.168.0.3 55380 typ host generation 0 network-id 1 network-cost 10
a=ice-ufrag:40h+
a=ice-pwd:6zJo50N7Bb2mqnrHq+jniukk
a=ice-options:trickle
a=fingerprint:sha-256 8A:F9:BE:8D:8A:80:FF:8C:89:3D:3A:D2:A1:36:B2:EC:11:53:81:7E:F4:53:E7:40:1E:B9:1E:A2:0F:D4:EA:2E
a=setup:active
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:oxU77z9z9RfNL4CayvM1cMJKI0r7u6ZdqLBd cb6da6b5-d5b8-460e-88bc-1458ebc718e6
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3489110087 cname:oZ2LjwJD385qPHHH

Kamailio handling replies using reply_route

onreply_route {
     if (nat_uac_test(64)) {
         add_contact_alias();
     }
 }

Sending ACK . 

ACK sip:john@voiptelco.com;alias=x.x.x.x~17510~6;gr=urn:uuid:2e560b36-3ea8-41fd-80e3-ede66babb8a7 SIP/2.0

Route: 
Via: SIP/2.0/WSS ipoct61ao12v.invalid;branch=z9hG4bK1876245
Max-Forwards: 69
To: sip:altanai@voiptelco.com ;tag=pvm73e3t89
From: sip:john@voiptelco.com;tag=2q0lecmbsn
Call-ID: s8bnv5869fp68d1ju8c1
CSeq: 1799 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.1.2
Content-Length: 0

since ACK is a Within dialog message and sequential request withing a dialog should take the path determined by record-routing , we first check if it has to tag. Having a to tag validates that it is a in-dialog request .

After this validate if is loose_route() and has no destination URI $du , then try to add rui alias using handle_ruri_alias() , if that fails , reject the request .

If it is not loose_route() and method is ACK then check if the ACK matches a transaction t_check_trans() ie is stateful . If it is then relay otherwise reject.

route(WITHINDLG);
...
route[WITHINDLG] {
	if (has_totag()) {
		if (loose_route()) {
			if ($du == "") {
				if (!handle_ruri_alias()) {
				xlog("L_ERR", "Bad alias <$ru&gt;\n");
				sl_send_reply("400", "Bad Request");
				exit;
				}
			}
			route(RELAY);
		} else {
			if ( is_method("ACK") ) {
				if ( t_check_trans() ) {
				t_relay();
				exit;
				} else {
				exit;
				}
			}
			sl_send_reply("404", "Not Found");
		}
		exit;
	}
}

Also required to convert ICE packet fromWebRTC to non ICE for Xlite.

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Posted in Kamailio

Tagged , ,

Session Border controller for WebRTC

Posted on August 2, 2016 | 2 comments

Unified communication services build around WebRTC should be vendor agnostic and multi-tenant and be supported by other Communication Service Providers (CSPs), SIP trunks, PBXs, Telecom Equipment Manufacturers (TEMs), and Communication Platform as a Service (CPaaS). This can happen if all endpoints adhere to SIP standards in most updated RFC. However since not all are on the boat , Session border controllers are a great way to mitigate the differences and provide seamless connectivity to signalling and media , which could be between WebRTC, SIP or PSTN, from TDM to IP .

Session Border Controllers ( SBC )  assist in controlling the signalling and usually also the media streams involved in calls and sessions.

They are often part of a VOIP network on the border where there are 2 peer networks of service providers such as backbone network and access network of corporate communication system which is behind firewall.

A more complex example is that of a large corporation where different departments have security needs for each location and perhaps for each kind of data. In this case, filtering routers or other network elements are used to control the flow of data streams. It is the job of a session border controller to assist policy administrators in managing the flow of session data across these borders. – wikipedia

SBC act like a SIP-aware firewall with proxy/B2BUA.

What is B2BUA?

A Back to back user agent ( B2BUA ) is a proxy-like server that splits a SIP transaction in two pieces:

  • on the side facing User Agent Client (UAC), it acts as server;
  • on the side facing User Agent Server (UAS) it acts as a client.

B2BUAs keep state information about active dialog. Read more here .

Remote Access

SBC mostly have public url address  for teleworkers and a internal IP for enterprise/ inner LAN . This enables users connected to enterprise LAN ( who do not have public address ) to make a call to user outside of their network. During this process SBC takes care of following while relaying packets .

  1. Security
  2. Connectivity
  3. Qos
  4. Regulatory
  5. Media Services
  6. Statistics and billing information

Topology hiding

SBC hides and anonymize secure information like IP ports before forwarding message to outside world . This helps protect the internal node of Operators such as PSTN gateways or SIP proxies from revealing outside.

Explaining the functions of SBC in detail

1. Security

SBCs are often used by corporations along with firewalls and intrusion prevention systems (IPS) to enable VoIP calls to and from a protected enterprise network. VoIP service providers use SBCs to allow the use of VoIP protocols from private networks with Internet connections using NAT, and also to implement strong security measures that are necessary to maintain a high quality of service. The security features includes :

  • Prevent malicious attacks on network such as DOS, DDos.
  • Intrusion detection
  • cryptographic authentication
  • Identity/URL based access control
  • Blacklisting bad endpoints
  • Malformed packet protection
  • Encryption of signaling (via TLS and IPSec) and media (SRTP)
  • Stateful signalling and Validation
  • Toll Fraud – detect who is intending to use the telecom services without paying up

2. Connectivity

As SBC offers IP-to-IP network boundary, it recives SIP request from users like REGISTER , INVITE  and routes them towards destination, making their IP. During this process it performs various operations like

  • NAT traversal
  • IPv4 to IPv6 inter-working
  • VPN connectivity
  • SIP normalization via SIP message and header manipulation
  • Multi vendor protocol normalization

Further Routing features includes  :
Least Cost Routing based on MoS ( Mean Opinion Score ) : Choosing a path based on MoS is better than chooisng any random path . 

Protocol translations between SIP, SIP-I, H.323.

In essence SBC achieve interoperability, overcoming some of the problems that firewalls and network address translators (NATs) present for VoIP calls.

Automatic Rerouting

connectivity loss from UA for whole branch is detected by timeouts . But they can also be detected by audio trough SIP OPTIONS by SBC .  In such connectivity loss , SBC decides rerouting or sending back 504 to caller .

SBC 2 (1)

4. QoS
To introduce performance optimization and business rules in call management QoS is very important . This includes the following :

  • Traffic policing
  • Resource allocation
  • Rate limiting
  • Call Admission Control (CAC)
  • ToS/DSCP bit setting
  • Recording and Audit of messages , voice calls , files
  • System and event logging

5. Regulatory

Govt policies ( such as ambulance , police ) and/ or enterprise policies may require some calls to be holding priority over others . This can also be configured under SBC as emergency calls and prioritization.
Some instances may require communication provider to comply with lawful bodies and provide session information or content , this is also called as Lawful interception (LI) . This enables security officials to collect specific information rather than examining all the traffic that passes through a particular router. This is also part of SBC.
6. Media services

Many of the new generation of SBCs also provide built-in digital signal processors (DSPs) to enable them to offer border-based media control and services such as- DTMF relay , Media transcoding , Tones and announcements etc.

WebRTC enabled SBC’s also provide conversion between DTLS-SRTP, to and from RTCP/RTP. Also transcoding for Opus into G7xx codecs
and ability to relay VP8/VP9 and H.264 codecs.

7. Statistics and billing information

SBC have an interface with and OSS/BSS systems for billing process , as almost all traffic that pass through the edge of the network passes via SBC. For this reason it is also used to gather Statistics and usage-based information like bandwidth, memory and CPU.  PCAP traces of both signaling and media information of specific sessions .

New feature rich SBCs also have built-in digital signal processors (DSPs). Thus able to provide more control over session’s media/voice . They also add services like Relay and Interworking, Media Transcoding, Tones and Announcements, DTMF etc.

Session Border Controller (SBC)
Session Border Controller for WebRTC , SIP , PSTN , IP PBX and Skype for business .

Diagram Component Description

Gateways provide compression or decompression, control signaling, call routing, and packetizing.

PSTN Gateway : Converts analog to VOIP and vice versa . Only audio no support for rich multimedia .

VOIP Gateway : A VoIP Gateway acts like a translator converting digital telecom lines to VoIP . VOIP gateway often also include voice and fax. They also have interfaces to Soft switches and network management systems.

WebRTC Gateway : They help in providing NAT with ICE-lite and STUN connectivity for peers behind policies and Firewall .

SIP trunking : Enterprises save on significant operation cost by switching to IP /SIP trunking in place of TDM (Time Division Multiplexing). Read more on SIP trunk and VPN  here. 

SIP Server : A Telecom application server ( SIP Server ) is useful for building VAS ( Value Added Services ) and other fine grained policies on real time services . Read more on SIP Servers here . 

VOIP/SIP service Provider :   There are many Worldwide SIP Service providers such as Verizon in USA , BT in europe, Swisscom in Switzerland etc .

Building a SBC

The latest trends in Telecommunications industry demand an open standardized SBC to cater to growing and large array of SIP Trunking, Unified Multimedia Communications UC&C, VoLTE, VoWi-Fi, RCS and OTT services worldwide . Building an SBC requires that it meet the following prime requirements :

  • software centric
  • Cloud Deploybale
  • Rich multimedia (audio , video , files etc) processing
  • open interfaces
  • The end product should be flexible to be deployed as COTS ( Commercial Off the shelf) product or as a virtual network function in the NFV cloud.
  • Multi Configuration , should be supported such as Hosted or Cloud deployed .
  • Overcome inconsistencies in SIP from different Vendors
  • Security and Lawful Interception
  • Carrier Grade Scaling

Flow Diagram 

SBC WebRTC to SIP

Thus we see how SBC became important part of comm systems developed over SIP and MGCP. SBC offer B2BUA ( Back to Back user agent) behavior to control both signalling and media traffic.

2 Comments

Posted in WebRTC SaaS

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Setting up ubuntu ec2 t2 micro for webrtc and socketio

Posted on May 17, 2016 | 2 comments

Setting up a ec2 instance on AWS for web real time communication platform over nodejs and socket.io using WebRTC .

Primarily a Web Call  , Chat and conference platform uses WebRTC for the media stream and socketio for the signalling . Additionally used technologies are nosql for session information storage , REST Apis foe getting sessions details to third parties.

Below is a comprehensive setup if ec2 t2.micro free tier instance  ,  installation with a webrtc project module and samples of customization and usuage .

Technologies used are listed below :

Server

  1. ec2 instance t2.micro covered under free tier
  2. domain name
  3. SSL certificate

Core module for Web Calling feature

  1. WebRTC
  2. Node.js
  3. socket.io

UI components

  1. javascript
  2. css
  3. html5
  4. bootstrap
  5. jquerry

Supporting setup for session management

  1. Code version-ing  and maintenance
  2. git
  3. npm

Amazon’s free tier ec2

Amazon EC2
ec2 instances are elastic compute general purpose storage servers that mean that they can resize the compute capacity in the cloud based on load .
750 hours per month of Linux, RHEL, or SLES t2.micro instance usage
Expires 12 months after sign-up.

Some other products are also covered under free tier which may come in handy for setting up the complete complatorm .Here is a quick summary

1.Amazon S3
it is a storage server. Can be used to store media file like image s, music , videos , recorded video etc .

2.Amazon RDS
It a relational database server . If one is using mysql or postgress for storing session information or user profile data . It is good option .

3.Amazon SES
email service. Can be used to send invites and notifications to users over mail for scheduled sessions or missed calls .

4.Amazon CloudFront
It is a CDN ( content delivery network ) . If one wants their libraries to be widly available without any overheads . CDN is a good choice .

Server Setup

Set up environment by installing nvm  , npm  and git ( source version control)

1. NVM ( node version manager )

cURL:

curl -o- https://raw.githubusercontent.com/creationix/nvm/v0.31.1/install.sh | bash

or Wget:

wget -qO- https://raw.githubusercontent.com/creationix/nvm/v0.31.1/install.sh | bash&amp;amp;amp;amp;amp;amp;amp;amp;amp;lt;/code&amp;amp;amp;amp;amp;amp;amp;amp;amp;gt;

To check installation

command -v nvm
nvm

2. NPM( node package manager)

sudo apt-get install npm

Screenshot from 2016-05-16 12-41-42

2. Git

sudo apt-get install git

Screenshot from 2016-05-17 11-25-01

 SSL certificates

Since 2015 it has become mandatory to have only https origin request WebRTC’s getUserMedia API ie Voice, video, geolocation , screen sharing require https origins.
Note that this does not apply to case where its required to only serve peer’s media Stream or using Datachannels . Voice, video, geolocation , screen sharing now require https origins

For A POC purpose here is th way of generating a self signed certificate
Transport Layer Security and/or Secure Socket Layer( TLS/SSL) is a public/private key infrastructure.Following are the steps

1.create a private key
openssl genrsa -out webrtc-key.pem 2048

2.Create a “Certificate Signing Request” (CSR) file
openssl req -new -sha256 -key webrtc-key.pem -out webrtc-csr.pem

3.Now create a self-signed certificate with the CSR,
openssl x509 -req -in webrtc-csr.pem -signkey webrtc-key.pem -out webrtc-cert.pem

However in production or actual implementation it is highly recommended to use a signed certificate by CA as For examples include
Godaddy (https://ca.godaddy.com/web-security/ssl-certificate) , Comoddo (https://ssl.comodo.com/) , Global Sign (https://www.globalsign.com/en/ssl/managed-ssl/) , Symantec (https://www.symantec.com/ssl-certificates) etc .

Web Server

create https certificate using self generate or purchased SSL certificates using fs , node-static and https modules . To know how to create self generated SSL certificates follow section above on SSL certificates.

var fs = require(‘fs’);
var _static = require(‘node-static’);
var https = require(‘https’);

var file = new _static.Server(&amp;amp;amp;amp;amp;amp;amp;quot;./&amp;amp;amp;amp;amp;amp;amp;quot;, {
cache: 3600,
gzip: true,
indexFile: &amp;amp;amp;amp;amp;amp;amp;quot;index.html&amp;amp;amp;amp;amp;amp;amp;quot;
});

var options = {
key: fs.readFileSync(‘ssl_certs/webrtc-key.pem’),
cert: fs.readFileSync(‘ssl_certs/webrtc-cert.pem’),
ca: fs.readFileSync(‘ssl_certs/webrtc-csr.pem’),
requestCert: true,
rejectUnauthorized: false
};

var app = https.createServer(options, function(request, response){
request.addListener(‘end’, function () {
file.serve(request, response);
}).resume();
});

app.listen(&amp;amp;amp;amp;amp;amp;amp;quot;8080&amp;amp;amp;amp;amp;amp;amp;quot;);

Web servers work with the HTTP (and HTTPS) protocol which is TCP based. As a genral rule TCP establishes connection whereas UDP send data packets

 

Scoketio signalling server as npm

Socket.io determines which of the following real-time communication method is suited to the particular client and its network bandwidth .

  • WebSocket
  • Adobe Flash Socket
  • AJAX long polling
  • AJAX multipart streaming
  • Forever Iframe
  • JSONP Polling

The socket.io server needs a HTTP Server for initial handshake.

The general steps for socketio signalling server are:

1.require socket.io and keep the reference. like
var io = require(‘socket.io’)

2.Create your http / https server
outline in section on webserver

3.bind your http and https servers (.listen)
io.listen(app, {
log: false,
origins: ‘*:*’
});

4. Optionally set transport
io.set(‘transports’, [
‘websocket’
]);

4.setup io events as
io.sockets.on(‘connection’, function (socket) {

//Do domething
});

Note that Socket.io or websockets require an http server for the initial handshake.
&amp;amp;lt;pre&amp;amp;gt;Install ssocketio npm module&amp;amp;lt;/pre&amp;amp;gt;&amp;amp;lt;pre&amp;amp;gt;
npm install socket.io
[/sourcecode ]

Complete code for signalling server

var io = require(‘socket.io’).listen(app, {
log: false,
origins: ‘*:*’
});

io.set(‘transports’, [
‘websocket’
]);

var channels = {};

io.sockets.on(‘connection’, function (socket) {

console.log(&amp;amp;amp;amp;amp;amp;amp;quot;connection &amp;amp;amp;amp;amp;amp;amp;quot;);
var initiatorChannel = ”;

if (!io.isConnected) {
io.isConnected = true;
}

socket.on(‘namespace’,function(data){
onNewNamespace(data.channel, data.sender);
});

socket.on(‘new-channel’, function (data) {
if (!channels[data.channel]) {
initiatorChannel = data.channel;
}
console.log(&amp;amp;amp;amp;amp;amp;amp;quot;————new channel &amp;amp;amp;amp;amp;amp;amp;quot;, data.channel , &amp;amp;amp;amp;amp;amp;amp;quot; by &amp;amp;amp;amp;amp;amp;amp;quot; , data.sender);
channels[data.channel] = {
channel: data.channel,
users:[data.sender]
};

});

socket.on(‘join-channel’, function (data) {
console.log(&amp;amp;amp;amp;amp;amp;amp;quot;————join&amp;amp;amp;amp;amp;amp;amp;amp;nbsp;channel &amp;amp;amp;amp;amp;amp;amp;quot;, data.channel , &amp;amp;amp;amp;amp;amp;amp;quot; by &amp;amp;amp;amp;amp;amp;amp;quot; , data.sender);
channels[data.channel].users.push(data.sender);
});

socket.on(‘presence’, function (channel) {
var isChannelPresent = !! channels[channel.channel];
console.log(&amp;amp;amp;amp;amp;amp;amp;quot;presence for channel &amp;amp;amp;amp;amp;amp;amp;quot; ,isChannelPresent);
socket.emit(‘presence’, isChannelPresent);
});

socket.on(‘disconnect’, function (channel) {
});

socket.on(&amp;amp;amp;amp;amp;amp;amp;quot;admin_enquire&amp;amp;amp;amp;amp;amp;amp;quot;,function(data){
switch (data.ask){
case &amp;amp;amp;amp;amp;amp;amp;quot;channels&amp;amp;amp;amp;amp;amp;amp;quot;:
socket.emit(‘response_to_admin_enquire’, channels);
break;
case &amp;amp;amp;amp;amp;amp;amp;quot;channel_clients&amp;amp;amp;amp;amp;amp;amp;quot;:
socket.emit(‘response_to_admin_enquire’, io.of(‘/’ + data.channel).clients());
break;
default :
socket.emit(‘response_to_admin_enquire’, channels);
}

});

});

function onNewNamespace(channel, sender) {
console.log(&amp;amp;amp;amp;amp;amp;amp;quot; —–&amp;amp;amp;amp;amp;amp;amp;amp;gt; onNewNamespace &amp;amp;amp;amp;amp;amp;amp;quot;, channel);

io.of(‘/’ + channel).on(‘connection’, function (socket) {

var username;
if (io.isConnected) {
io.isConnected = false;
socket.emit(‘connect’, true);
}

socket.on(‘message’, function (data) {
if (data.sender == sender) {
if(!username) username = data.data.sender;
socket.broadcast.emit(‘message’, data.data);
}
});

socket.on(‘disconnect’, function() {
if(username) {
socket.broadcast.emit(‘user-left’, username);
username = null;
}
});
});
}

 

WebRTC main HTML5  project

This is the front  end section of the whole exercise . It contains JavaScript , css and html5 to make a webrtc call

<html lang=en>
<head>
<title>WebRTC Call</title>

<meta http-equiv=Content-Type content="text/html; charset=UTF-8">
<meta name="viewport" content="width=device-width, initial-scale=1">

	<link rel=stylesheet href="https://ec2-54-193-124-35.us-west-1.compute.amazonaws.com:8084/minScripts/webrtcdevelopment_header.css">
<script src="https://ec2-54-193-124-35.us-west-1.compute.amazonaws.com:8084/minScripts/webrtcdevelopment_header.js"> </script>

<style type="text/css">
video{
width:100% !important;
}
body{
background: #2B2B2B;
}
</style>
</head>

<body id="pagebody">
<div id="elementToShare" class="container-fluid">
<!-- ................................ top panel ....................... -->
<div class="row topPanelClass" >
<div id="topIconHolder" >
<ul id="topIconHolder_ul">
	<li hidden> <span id="username" class="userName" hidden>a</span></li>
	<li hidden> <span id="numbersofusers" class="numbers-of-users" hidden></span></li>
	<li> <span id="HelpButton" class="btn btn-info glyphicon glyphicon-question-sign topPanelButton" data-toggle="modal" data-target="#helpModal" > Help </span></li>
</ul>
</div>
</div>
<!-- .............alerts................. -->
<div class="row" id="alertBox" hidden="true"></div>
<!-- .......................... Row ................................ -->
<div class="row thirdPanelClass">
<div class="col-xs-12 videoBox merge" id="videoHold">
<div class="row users-container merge" id="usersContainer" >
<div class="CardClass" id="card">

<!-- when no remote -->
<div id="local" class="row" hidden="">
<video name="localVideo" autoplay="autoplay" muted="true" />
</div>
<!-- when remote is connected -->
<div id ="remote" class="row" style="display:inline" hidden>
<div class="col-sm-6 merge" class="leftVideoClass" id="leftVideo">
<video name="video1" hidden autoplay="autoplay" muted="true" ></video>
</div>
<div class="col-sm-6 merge" class="rightVideoClass" id="rightVideo" >
<video name="video2" hidden autoplay="autoplay" ></video>
</div>
</div>
</div>
</div>
</div>
</div>
<!--modal help -->
<div class="modal fade" id="helpModal" role="dialog">
<div class="modal-dialog modal-lg">
<div class="modal-content">
<div class="modal-header">
<button type="button" class="close" data-dismiss="modal">&times;</button>
<h4 class="modal-title">Help</h4>
</div>
<div class="modal-body">
WebRTC Runs in only https due to getusermedia security contraints
</div>
<div class="modal-footer">
<button type="button" class="btn btn-default" data-dismiss="modal">Close</button>
</div>
</div>
</div>
</div>
</div>
</body>

	<link rel=stylesheet href="https://ec2-54-193-124-35.us-west-1.compute.amazonaws.com:8084/minScripts/webrtcdevelopment.css">
<script src="https://ec2-54-193-124-35.us-west-1.compute.amazonaws.com:8084/minScripts/webrtcdevelopment.js"> </script>

<script>
$('document').ready(function(){

 sessionid= init(true);

 var local={
 localVideo: "localVideo",
 videoClass:"",
 userDisplay:false,
 userMetaDisplay:false 
 };

 var remote={
 remotearr: ["video1" , "video2"],
 videoClass:"",
 userDisplay:false,
 userMetaDisplay:false 
 };

 webrtcdomobj= new WebRTCdom(
 local,remote
 );

 var session ={
 sessionid : sessionid,
 socketAddr: "https://localhost:8084/"
 };

 var webrtcdevobj = new WebRTCdev ( session, null, null , null );

 startcall();
});
</script>
</html>

Screenshot from 2016-05-17 12-12-37.png

Common known issues:

1.Opening page https://<web server ip>:< web server port>/index.html says insecure

This is beacuse the self signed certificates produced by open source openSSL is not recognized by a trusted third party Certificate Agency.
A CA ( Certificate Authority ) issues digital certificate to certify the ownership of a public key for a domain.

To solve the access issue goto https://<web server ip>:< web server port> and given access permission such as outlined in snapshot below

image

2.Already have given permission to Web Server , page loads but yet no activity .

if you open developer console ( ctrl+shift+I on google chrome ) you will notice that there migh be access related errros in red .
If you are using different server for web server and signalling server or even if same server but different ports you need to explicity go to the signalling server url and port and give access permission for the same reason as mentione above.
3.no webcam capture on opening the page

This could happen due to many reasons

  •  page is not loaded on https
  • browser is not webrtc compatible
  • Media permission to webcam are blocked
  • the machine does have any media capture devices attached
  •  Driver issues in the client machine while accessing webcams and mics .

4.socketio + code: 0, message: “Transport unknown”

Due to the version  v1.0.x of socket.io while performing handshake . To auto correct this , downgrade to v0.9.x

 

 

2 Comments

Posted in Telecom Info, WebRTC SaaS

Tagged , , , , ,

RamuDroid

Posted on March 18, 2016 | Leave a comment

Bot to clean roads and outdoors for a better and cleaner India. It lifts up small objects like plastic cups,wrappers,leaves etc.

ramudroid image.png

The droid also provides real-time camera stream and detects obstruction to re-route itself. It can communicates over GSM ,wifi and BLE . It can also be remote navigated via browsers or android.

Working :

stages of ramudroid

  1. Litter comes between rotating brushes
  2. Litter is picked by brushes and pushed upwards  
  3. Brushes push it towards the tray

Is is inspired by Swach Bharat Abhiyaan in India , its an effort to contribute to society and welfare and well-being through technology . Following are some diagrams for the current and the previous versions , along with major delta points .

RamuDroid v1.0

Remote Streaming and movement via motors switched manually. Communication over Ethernet.

Ramudroidv1.0.jpg

Dashboard /console Screen

RamuDroid v1 console

v3.0

Cleaning garbage on public roads and outdoors through robot . Remote navigation and control of control through web page and camera live streaming .

Ramudroid compoenet diagram v5

v3 web console

Screenshot from 2015-12-03 08:55:27.png

v6

Clean roads , pick up litter ( wrappers, leaves , cups , plastics bits etc ) . communicated over BLE ,Wifi and 3G n/w . Auto buzzer when meet with an obstruction in way . Flash Lights . Enhanced Design .

Ramudroid 6.5 componet diagram

 

Web Dashboard:

Screenshot from 2016-03-19 04-28-53.png

Pin Diagram associated with activities

Pin Pin 0 Pin 1 Pin 2 Pin 3 Pin 4 Pin 5 Pin 6 Pin 7
Front 0 1 0 1 1 1 1 1
Back 1 0 1 0 1 1 1 1
Left 1 0 1 0 1 1 1 1
Right 1 0 1 0 1 1 1 1
Brushes ON 1 0 1 0 1 1 1 1
Brushes OFF 1 0 1 0 1 1 1 1
Lift ON 1 0 1 0 1 1 1 1
Life OFF 1 0 1 0 1 1 1 1

 

Github : https://github.com/altanai/m2mcommunication

Slideshare :

Twitter :https://twitter.com/search?q=%23ramudroid

 

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Posted in Robotics

Tagged , , , , , ,

WebGL , Three.js and WebRTC

Posted on November 11, 2015 | 2 comments

For the last couple of weeks , I have been working on the concept of rendering 3D graphics on WebRTC media stream using different JavaScript libraries as part of a Virtual Reality project .

What is Augmented Reality ?

Augmented reality (AR) is viewing a real-world environment with elements that are supplemented by computer-generated sensory inputs such as sound, video, graphics , location etc.

How is it diff. from Virtual Reality ?

Virtual Reality – replaces the real world with simulated one , user is isolated from real life , Examples – Oculus Rift & Kinect

Augmented Reality – blending of virtual reality and real life , user interacts with real world through digital overlays , Examples – Google glass & Holo Lens

Methods for rendering augmented Reality

  • Computer Vision
  • Object Recognition
  • Eye Tracking
  • Face Detection and substitution
  • Emotion and gesture picker
  • Edge Detection

web based Augmented Reality solution

Components for a Web base  end to end AR solution

Web :

WebRTC getusermedia
Web Speech API
WebGL
css
svg
HTML5 canvas
sensor API

H/w :

Graphics driver
microphone and camera
sensors

3D :

Geometry and Math Utilities
3D Model Loaders and models
Lights, Materials,Shaders, Particles,
Animation

WebRTC

  • Web based Real Time communications
  • Definition for browser’s media stream and data
  • Awaiting more standardization , on a API level at the W3C and at the protocol level at the IETF.
  • Enable browser to browser applications for voice calling, video chat and P2P file sharing without plugins.
  • Enables web browsers with Real-Time Communications (RTC) capabilities
  • MIT : Free, open project

Code snippet for WebRTC API

1.To begin with WebRTC we first need to validate that the browser has permission to access the webcam.

&amp;amp;lt;video id=&amp;amp;quot;webcam&amp;amp;quot; autoplay width=&amp;amp;quot;640&amp;amp;quot; height=&amp;amp;quot;480&amp;amp;quot;&amp;amp;gt;&amp;amp;lt;/video&amp;amp;gt;
  1. Find out if the user’s browser can use the getUserMedia API.
function hasGetUserMedia() {
	return !!(navigator.webkitGetUserMedia);
}
  1. Get the stream from the user’s webcam.
var video = $('#webcam')[0];
if (navigator.webkitGetUserMedia) {
         navigator.webkitGetUserMedia(
			{audio:true, video:true},
			function(stream) { video.src = window.webkitURL.createObjectURL(stream);  },
			function(e) {	alert('Webcam error!', e); }
		);
}

Screenshot AppRTC

https://apprtc.appspot.com

Augmented Reality in WebRTC Browser - Google Slides

 

WebGL

  • Web Graphics Library
  • JavaScript API for rendering interactive 2D and 3D computer graphics in browser
  • no plugins
  • uses GPU ( Graphics Processing Unit ) acceleration
  • can mix with other HTML elements
  • uses the HTML5 canvas element and is accessed using Document Object Model interfaces
  • cross platform , works on all major Desktop and mobile browsers

WebGL Development

To get started you should know about :

  • GLSL, the shading language used by OpenGL and WebGL
  • Matrix computation to set up transformations
  • Vertex buffers to hold data about vertex positions, normals, colors, and textures
  • matrix math to animate shapes

Cleary WebGL is bit tough given the amount of careful coding , mapping and shading it requires .

Proceeding to some JS libraries that can make 3D easy for us .

CCV

Three.js

Awe.js

ArcuCO

Potree

Karenpeng

Three.JS

  • MIT license
  • javascript 3D engine ie ( WebGL + more)
  • started a year ago
  • under active development
  • no dependencies or installation

3D space with webcam input as texture

Display the video as a plane which can be viewed from various angles in a given background landscape. Credits for code : https://stemkoski.github.io/Three.js/

1.Use code from slide 10 to get user’s webcam input through getUserMedia

  1. Make a Screen , camera and renderer as previously described

  2. Give orbital CONTROLS for viewing the media plane from all angles

	controls = new THREE.OrbitControls( camera, renderer.domElement );
  1. Add point LIGHT to scene

  2. Make the FLOOR with an image texture

	var floorTexture = new THREE.ImageUtils.loadTexture( 'imageURL.jpg' );
	floorTexture.wrapS = floorTexture.wrapT = THREE.RepeatWrapping;
	floorTexture.repeat.set( 10, 10 );
	var floorMaterial = new THREE.MeshBasicMaterial({map: floorTexture, side: THREE.DoubleSide});
	var floorGeometry = new THREE.PlaneGeometry(1000, 1000, 10, 10);
	var floor = new THREE.Mesh(floorGeometry, floorMaterial);
	floor.position.y = -0.5;
	floor.rotation.x = Math.PI / 2;
	scene.add(floor);</pre>
<pre>

6. Add Fog


scene.fog = new THREE.FogExp2( 0x9999ff, 0.00025 );

7.Add video Image Context and Texture.

 
video = document.getElementById( 'monitor' ); 
videoImage = document.getElementById( 'videoImage' ); 
videoImageContext = videoImage.getContext( '2d' ); 
videoImageContext.fillStyle = '#000000'; 
videoImageContext.fillRect( 0, 0, videoImage.width, videoImage.height ); 
videoTexture = new THREE.Texture( videoImage ); 
videoTexture.minFilter = THREE.LinearFilter; 
videoTexture.magFilter = THREE.LinearFilter; 
var movieMaterial=new THREE.MeshBasicMaterial({map:videoTexture,overdraw:true,side:THREE.DoubleSide}); 
var movieGeometry = new THREE.PlaneGeometry( 100, 100, 1, 1 ); 
var movieScreen = new THREE.Mesh( movieGeometry, movieMaterial ); 
movieScreen.position.set(0,50,0);
scene.add(movieScreen);
  1. Set camera position
	camera.position.set(0,150,300);
	camera.lookAt(movieScreen.position);
  1. Define the render function
    videoImageContext.drawImage( video, 0, 0, videoImage.width, videoImage.height );
    renderer.render( scene, camera );
  1. Animation
   requestAnimationFrame( animate );
   render();

 

Augmented Reality in WebRTC Browser - Google Slides (4)

 

 

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Posted in Augmented Reality

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WebRTC Live Stream Broadcast

Posted on August 7, 2015 | 1 comment

WebRTC has the potential to drive the Live Streaming broadcasting area with its powerful no plugin , no installation , open standard  policy . However the only roadblock is the VP8 codec which differs from the traditional H264 codec that is used by almost all the media servers , media control units , etc .

This post is first in the series of building a WebRTC based broadcasting solution. Note that a p2p session differs from a broadcasting session as Peer-to-peer session applies to bidirectional media streaming where as broadcasting only applies unidirectional media flow.

Scalable Broadcasting and Live streaming alternatives

1. WebRTC multi peers

Since WebRTC is p2p technology , it is convenient to build a  network of webrtc client viewers which can pass on the stream to 3 other peers in different session. In this fashion a fission chain like structure is created where a single stream originated to first peer is replicated to 3 others which is in turn replicated to 9 peers etc .

WebRTC Scalable Streaming Server -WebRTC multi peers

WebRTC Scalable Streaming Server -WebRTC multi peers

Advantages :

  1. Scalable without the investment of media servers
  2. No additional space required at service providers network .

Disadvantage :

  1. The entire set of end clients to a node get disconnected if a single node is broken .
  2. Since sessions are dynamically created , it is difficult to maintain a map with fallback option in case of service disruption from any single node .
  3. Client incur bandwidth load of 2 Mbps( stream incoming peer ) incoming and 6 Mbps ( for 3 connected peers ) outgoing data .

2. Torrent based WebRTC chain

To over come the shortcoming of previous approach of  tree based broadcasting , it is suggested to use a chained broadcasting mechanism .

WebRTC Scalable Streaming Server v1 (4)

WebRTC Scalable Streaming Server- Single chain connection

To improvise on this mechanism for incresing efficieny for slow bandwidth connections we can stop their outgoing stream converting them to only consumers . This way the connection is mapped and arranged in such a fashion that every alternate peer is connected to 2 peers  for stream replication. The slow bandwidth clients can be attached as independent endpoints . WebRTC Scalable Streaming Server v1 (3)

3. WebRTC Relay nodes for multiple peers

The aim here is to build a career grade WebRTC stream broadcasting platform , which is capable of using the WebRTC’s mediastream and peerconnection API , along with repeaters to make a scalable broadcasting / live streaming solution using socketio for behavior control and signalling .

Algorithm :

At the Publisher’s end

1. GetUserMedia
2. Start Room “liveConf”
3. Add outgoing stream to session “liveConf “ with peer “BR” in 1 way transport .

1 outgoing audio stream -> 1 MB in 1 RTP port
1 outgoing video -> 1 MB 1 more RTP port
Total Required 2 MB and 2 RTP ports

At the Repeater layer (high upload and download bandwidth )

4. Peer “BR” opens parallel room “liveConf_1” , “liveConf_2” with 4 other peers “Repeater1 “, “Repeater2” , so on
5. Repetare1 getRemoteStream from “liveConf_1” and add as localStream to “liveConf_1_1”

Here the upload bandwidth is high and each repeater is capable of handling 6 outgoing streams . Therefore total 4 repeaters can handle upto 24 streams very easily

At the Viewer’s end

6. Viewer Joins room ”liveConf_1_1”
7. Play the incoming stream on WebRTC browser video element”

WebRTC Relay nodes for multiple peers

WebRTC Relay nodes for multiple peers

Advantages:

  1. As 6 viewers can connect to 1 repeater for feed , total of 24 viewers will require only 4 repeaters.
  2. Only 2 MB consumption at publishers end and 2MB at each viewer’s end.

4. WebRTC  recorder to Broadcasting Media Server VOD

This process is essentially NOT a live streaming solution but a Video On Demand type of implementation for a recorded webRTC stream .

Figure shows a WebRTC node which can record the webrtc files as webm . Audio and video can be together  recorded on fireox . With chrome one needs to merge a separately recorded webm ( video) and wav ( audio ) file to make a single webm file containing both audio and video and them store in VOD server’s repo.

WebRTC Scalable Streaming Server - WebRTC Chunk recorder to Broadcasting Media Server VOD

WebRTC Chunk recorder to Broadcasting Media Server VOD

Although inherently Media Server do not support webm format but few new age lightweight media servers such as Kurento are capable of this .

Advantages :

  1. Can solve the end goal of broadcasting from a webrtc browser to multiple webrtc browsers without incurring extra load on any client machine ( Obviously assuming that  Media Server handles the distribution of video and load sharing automatically )

Disadvantages:

  1. It is not livestreaming
  2. For significantly longer recorded stream the delta in delay of streaming increases considerably .  Ideally this delta should be no more than 5 minutes .

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Posted in Live Streaming and Broadcasting

Tagged , , , , ,

TFX Widgets Development

Posted on May 19, 2015 | Leave a comment

TFX is a modular widget based WebRTC communication and collaboration solution. It is a customizable solution where developers can create and add their own widget over the underlying WebRTC communication mechanism . It can support extensive set of user activity such as video chat , message , play games , collaborate on code , draw something together etc . It can go as wide as your imagination . This post describes the process of creating widgets to host over existing TFX platform .

Prerequisites

It is required to have TFX Chrome extension installed and running from Chrome App Store under above . To do this follow the steps described in TangoFX v0.1 User’s manual.

Test TFX Sessions  ?

TFX Sessions uses the browser’s media API’s , like getUserMedia and Peerconnection to establish p2p media connection . Before media can traverse between 2 end points the signalling server is required to establish the path using Offer- Answer Model . This can be tested by making unit test cases on these function calls .

TFX Sessions uses socketio based handshake between peers to ascertain that they are valid endpoints to enter in a communication session . This is determined by SDP ( Session Description Parameters ) . The same can be observed in chrome://webrtc-internals/ traces and graphs .

TangoFX v8 Developer's manual - Google Docs

How to make widgets using TFX API ?

Step 1:  To make widgets for TFX , just write your simple web program which should consist of one main html webpage and associated css and js files for it .

Step 2 : Find an interesting idea which is requires minimal js and css . Remember it is a widget and not a full fleshed web project , however js frameworks like requirejs , angularjs , emberjs etc , work as well.

Step 3: Make a compact folder with the name of widget and put the respective files in it. For example the html files or view files would go to src folder , javascript files would goto js folder , css files would goto css folder , pictures to picture folder , audio files to sound folder and so on .

Step 4 : Once the widget is performing well in standalone environment , we can add a sync file to communicate the peer behaviors across TFX network . For this we primarily use 2 methods .

  • SendMessage : To send the data that will be traversed over DataChannel API of TFX . The content is in json format and will be shared with the peers in the session .
  • OnMessage : To receive the message communicated by the TFX API over network

Step 5: Submit the application to us or test it yourself by adding the plugin description in in widgetmanifest.json file . Few added widgets are

[
{
"plugintype": "code",
"id" 	:"LiveCode",
"type" 	: "code",
"title" : "Code widget",
"icon" 	: "btn btn-style glyphicon glyphicon-tasks", 
"url"	: "../plugins/AddCode/src/codewindow.html"
},

{
"plugintype": "draw",
"id" 	:"Draw",
"type" 	: "draw",
"title" : "Code widget",
"icon" 	: "btn btn-style glyphicon glyphicon-pencil", 
"url"	: "../plugins/AddDraw/src/drawwindow.html"
},

{
"plugintype": "pingpong",
"id"   :"Pingpong",
"type" : "pingpong",
"title": "ping pong widget",
"icon" : "btn btn-style glyphicon glyphicon-record", 
"url"  : "../plugins/AddPingpong/src/main.html"
}
]

Step 6 : For proper orientation of the application make sure that overflow is hidden and padding to left is atleast 60 px so that it doesnt overlap with panel
padding-left: 60px;
overflow: hidden;

Step 7 : Voila the widget is ready to go .

Simple Messaging Widget

For demonstration purpose I have summarised the exact steps followed to create the simple messaging widget which uses WebRTC ‘s Datachannel API in the back and TFX SendMessage & OnMessage API to achieve

Step 1 : Think of a general chat scenario as present in various messaging si

Step 2: Made a folder structure with separation for js , css and src. Add the respective files in folder. It would look like following figure:

TangoFX v8 Developer's manual - Google Docs (1)
Step 3 : The html main page is

<html>
<head>
<!-- jquerry -->
<script src="../../../../js/jquery/jquery.min.js"></script>
<link rel="stylesheet" type="text/css" href="../../../../css/jquery-ui.min.css"/>
<link rel="stylesheet" type="text/css" href="../css/style.css">
</head>

<body>
<div id="messages" class="message_area">
<textarea disabled class="messageHistory" rows="30" cols="120" id="MessageHistoryBox"></textarea>
   <br/>
<input type ="text" id="MessageBox" size="120"/>
</div>
</body>
<script src="../js/sync.js" type="text/javascript"></script>
</html>

Step 4 : The contents of css file are

body {padding: 0; margin: 0; overflow: hidden;}

.message_area{
	  padding-left: 60px;
}
.messageHistory{
	background-color: transparent;
	background: transparent;
}

Step 5 : The contents of js file

//send message when mouse is on mesage dicv ans enter is hit
$("#messages").keyup(function(event){
    if(event.keyCode == 13){
    var msg=$('#MessageBox').val(); 
    //send to peer 
    var data ={
       "msgcontent":msg
      }
    sendMessage(data);
    addMessageLog(msg);
    $("#MessageBox").val('');
    }
});

function addMessageLog(msg){
    //add text to text area for message log for self 
 $('#MessageHistoryBox').text( $('#MessageHistoryBox').text() + '\n'+ 'you : '+ msg);
}

// handles send message
function sendMessage(message) {
      var widgetdata={
      "type":"plugin",
      "plugintype":"relaymsg",
      "action":"update",
      "content":message
      };
  // postmessage
  window.parent.postMessage(widgetdata,'*');
}

//to handle  incoming message
function onmessage(evt) {
    //add text to text area for message log from peer
  if(evt.data.msgcontent!=null ){
      $('#MessageHistoryBox').text( $('#MessageHistoryBox').text() +'\n'+ 'other : '+ evt.data.msgcontent );
  }
}

window.addEventListener("message",onmessage,false);

Step 6: The end result is :

TangoFX v8 Developer's manual - Google Docs (2)

Developing a cross origin Widget ( XHR)

Let us demonstrate the process and important points to create a cross- origin widget :

step 1 : Develop a separate web project and run it on a https

step 2 : Add the widget frame in TFX . Following is the code I added to make an XHR request over GET

var xmlhttp;
xmlhttp=new XMLHttpRequest();
xmlhttp.onreadystatechange=function()
  {
  if (xmlhttp.readyState==4 && xmlhttp.status==200)
    {
    document.getElementById("myDiv").innerHTML=xmlhttp.responseText;
    }
  }

xmlhttp.open("GET","https://192.168.0.119:8000/TFXCrossSiteProj/files/document1.txt",true);
xmlhttp.send();

step 3 : Using self made https we have have to open the url separately in browser and give it explicit permission to open in advanced setting. Make sure the original file is visible to you at the widgets url .

TangoFX v8 Developer's manual - Google Docs (3)
step 4: Adding permission to manifest for access the cross origin requests

"permissions": [
  "tabs","http://*.google.com/",
  "https://192.168.0.119:8000/TFXCrossSiteProj"
],

Step 5 : Rest of the process are similar to develop a regular widget ie css and js .

Step 6: Resulting widget on TFX

TangoFX v8 Developer's manual - Google Docs (4)

Note 1 :
In absence of changes to manifest file the cross origin request is meet with a Access-Control-Allow-Origin error .

Note 2:
While using POST the TFX responds with Failed to load resource: the server responded with a status of 404 (Not Found)

Note 3:
Also if instead of https http is used the TFX still responds with Failed to load resource: the server responded with a status of 404 (Not Found)

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TURN server for WebRTC – RFC5766-TURN-Server , Coturn , Xirsys

Posted on March 28, 2015 | 2 comments

STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) are protocols that can be used to provide NAT traversal for VoIP and WebRTC. These projects provide a VoIP media traffic NAT traversal server and gateway.

TURN Server is a VoIP media traffic NAT traversal server and gateway.

I come accross the question of difference between turn and stun a lot . Here I wanted to specify in very clear words that TURN is an extension of STUN .

rfc5766-turn-server

This is a VoIP gateway for inter network communication which is popular and MIT based .

platforms supported :

Any client platform is supported, including Android, iOS, Linux, OS X, Windows, and Windows Phone. This project can be successfully used on other *NIX platforms ( Aamazon EC2) too. It supports flat file or Database based user management system ( MySQL , postgress , redis ). The source code project contains ,  TURN server ,  TURN client messaging library and some sample scripts to test various modules like protocol , relay , security etc .

Protocols :

protocols between the TURN client and the TURN server – UDP, TCP, TLS, and DTLS. Relay protocol – UDP , TCP .

Authentication

The authentication mechanism is using key which is calculated over the user name, the realm, and the user password. Key for the HMAC depends on whether long-term or short-term credentials are in use. For long-term credentials, the key is 16 bytes:
key = MD5(username “:” realm “:” SASLprep(password))

Installation

Since I used my Ubuntu Software center for installing the RFC turn server 5677 .

Screenshot from 2015-03-05 15:22:30

More information is on Ubuntu Manuals : http://manpages.ubuntu.com/manpages/trusty/man1/turnserver.1.html

The content got stored inside /usr/share/rfc5766-turn-server.

Also install mysql for record keeping

sudo apt-get install mysql-server

mysql

mysql2mysql4

Intall MySQL workbench to monitor the values feed into the turn database server in MysqL. connect to MySQL instance using the following screenshot

mysql5

The database formed with mysql after successful operation is as follows . We  shall notice that the initial db is absolutely null

mysql8empty

Terminal Commands

These terminal command ( binary images ) get stored inside etc/init.d after installing

turnadmin –

Its turn relay administration tool used for generating , updating keys and passwords . For generating a key to get long term crdentaial use -k command and for aading or updateing a long -term user use the -a command. Therefore a simple command to generate a key is

format : turnadmin -k -u -r -p
examples : turnadmin -k -u turnwebrtc -r mycompany.com -p turnwebrtc

The generated key is displayed in console . For example the following screenshot shows this :

rfc5677turnkey

To fill in user with long term credentails

Format : turnadmin -a [-b | -e | -M | -N ] -u -r -p

exmaple : turnadmin -a -M “host=localhost dbname=turn user=turn password=turn” -u altanai -r mycompany.com -p 123456

Check the values reflected in MySQL workbench for long term user table . ( screenshot depicts two entries for altanai and turnwebrtc user )

turnkeylongterm

you can also check it on console using the -l command

format :turnadmin -l –mysql-userdb=””

example :  turnadmin -l –mysql-userdb=”host=127.0.0.1 dbname=turn user=turnwebrtc password=turnwebrtc connect_timeout=30″

longtermuserlcommand

or we can also check using the terminal based mySQL client

mysql> use turn;
Database changed

mysql> select * from turnusers_lt;
+------------+----------------------------------+
| name | hmackey |
+------------+----------------------------------+
| altanai | 57bdc681481c4f7626bffcde292c85e7 |
| turnwebrtc | 6066cbe0b5ee14439b2ddfc177268309 |
+------------+----------------------------------+
2 rows in set (0.00 sec)

turnserver –

Its command to handle the turnserver itself . We can use the simple turnserver command to start it without any db support using just turnserver. Screenshot for this is

turnserverstart

We can use a database like mysql to start it with db connection string

Format : turnserver –mysql-userdb=””

Example : turnserver –mysql-userdb=”host=127.0.0.1 dbname=turn user=turnwebrtc password=turnwebrtc connect_timeout=30″

turnservermysqldb

turnutils_uclient:

emulates multiple UDP,TCP,TLS or DTLS clients.

turnutils_peer:

simple stateless UDP-only “echo” server. For every incoming UDP packet, it simply echoes it back.

turnutils_stunclient:

simple STUN client example that implements RFC 5389 ( using STUN as endpoint to determine the IP address and port allocated to it , keep-alive , check connectivity etc) and RFC 5780 (experimental NAT Behavior Discovery STUN usage) .

turnutils_rfc5769check:

checks the correctness of the STUN/TURN protocol implementation. This program will perform several checks and print the result on the screen. It will exit with 0 status if everything is OK, and with (-1) if there was an error in the protocol implementation.

Specifications :

TURN specifications include :

  • RFC 5766 – base TURN specs
  • RFC 6062 – TCP relaying TURN extension
  • RFC 6156 – IPv6 extension for TURN
  • DTLS
  • Mobile ICE (MICE)

STUN specifications :

  • RFC 3489 – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs) (STUN) to discover the presence and public IP
  • RFC 5389 – STUN serves as a tool for other protocols in NAT traversal. It can be used by an endpoint to determine the IP address and port allocated to it , keep-alive  , check connectivity etc .
  • RFC 5769 – test vectors for STUN protocol . FINGERPRINT, MESSAGE-INTEGRITY, and XOR-MAPPED-ADDRESS involving binary-logical operations (hashing, xor)
  • RFC 5780 – experimental NAT Behavior Discovery STUN usage

ICE specifications :

  • RFC 5245 – ICE
  • RFC 5768 – ICE–SIP
  • RFC 6336 – ICE–IANA Registry
  • RFC 6544 – ICE–TCP
  • RFC 5928 – TURN Resolution Mechanism

Test :

1. Test vectors from RFC 5769 to double-check that our
STUN/TURN message encoding algorithms work properly. Run the utility to check all protocols :

$ cd examples
$ ./scripts/rfc5769.sh

2. TURN functionality test (bare minimum TURN example).

If everything compiled properly, then the following programs must run
together successfully, simulating TURN network routing in local loopback
networking environment:

console 1 :

$ cd examples
$ ./scripts/basic/relay.sh

console2 :

$ cd examples
$ ./scripts/peer.sh

If the client application produces output and in approximately 22 seconds
prints the jitter, loss and round-trip-delay statistics, then everything is
fine.

Usage

iceServers:[
{ ‘url’: ‘stun: altanai@mycompany.com’},
{ ‘url’: ‘turn: altanai@mycompany.com’, ‘credential’: ‘123456’}]

Insert the above piece of code on peer connection config .

Now call from one network environment to another . For example call from a enterprise network behind a Wifi router to a public internet datacard webrtc agent . The call should connect with video flowing smoothly between the two .

tooltips

website : https://code.google.com/p/rfc5766-turn-server/

Download the executable from : http://turnserver.open-sys.org/downloads/v3.2.5.4/

you can read about setting a carrier grade TURN infrastructure on amazon EC2 here –

coturn

Project Coturn evolved from rfc5766-turn-server project with many new advanced TURN specs beyond the original RFC 5766 document.
Here the databses supported are : SQLite , MySQL , PostgreSQL , Redis , MongoDB

Protocols :

The implementation fully supports the following client-to-TURN-server protocols: UDP  , TCP  , TLS  SSL3/TLS1.0/TLS1.1/TLS1.2; ECDHE , DTLS versions 1.0 and 1.2. Supported relay protocols UDP (per RFC 5766) and TCP (per RFC 6062)

Authetication :

Supported message integrity digest algorithms:

  • HMAC-SHA1, with MD5-hashed keys (as required by STUN and TURN standards)
  • HMAC-SHA256, with SHA256-hashed keys (an extension to the STUN and TURN specs)

Supported TURN authentication mechanisms:

Installation :

Install libopenssl and libevent plus its dev or extra libraries .
OpenSSL has to be installed before libevent2 for TLS beacuse When libevent builds it checks whether OpenSSL has been already installed, and its version.

Download coturn readonly  from

svn checkout http://coturn.googlecode.com/svn/trunk/ coturn-read-only

extract the tar contents
$ tar xvfz turnserver-.tar.gz

go inside the extracted folder and run the following command to build
$ ./configure
$ make
$ make install

Adding users in the format using turnadmin
$ Sudo turnadmin -a -u -r -p

Example
$ Sudo turnadmin -a -u altanai -r myserver.com -p 123456

Start the turn Server using turnserver from inside of /etc/init.d using the start command

$ sudo /etc/init.d/coturn start

Screenshot from 2015-01-06 12-08-15

The logs are usually stored in /var/log . Screenshot of log file

tuenlog2

The default configured port is 3478.If other port is needed, change the file /etc/turnserver.conf

Usuage:

Specify the  values in Peer Connection

Format:
iceServers: [
{ ‘url’: ‘stun: @: ‘},
{ ‘url’: ‘turn: @: ‘, ‘credential’: ”}]

example:

iceServers:[
{ ‘url’: ‘stun: altanai@myserver.com’},
{ ‘url’: ‘turn: altanai@myserver.com’, ‘credential’: ‘123456’}]

Specifications:

TURN specs:

STUN specs:

  • RFC 3489 – STUN – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
  • RFC 5389 – Session Traversal Utilities for NAT (STUN)
  • RFC 5769 – test vectors for STUN protocol testing
  • RFC 5780 – NAT behavior discovery support
  • RFC 7443 – Application-Layer Protocol Negotiation (ALPN) Labels for STUN and TURN

ICE :

  • RFC 5245 – ICE
  • RFC 5768 – ICE–SIP
  • RFC 6336 – ICE–IANA Registry
  • RFC 6544 – ICE–TCP
  • RFC 5928 – TURN Resolution Mechanism

website : https://code.google.com/p/coturn/

Xirsys

Xirsys is a provider for WebRTC infrastructure which included stun and turn server hosting as well .

The process of using their services includes singing up for a account and choosing whether you want a paid service capable of handling more calls simultaneously or free one handling only upto 10 concurrent turn connections .

The dashboard appears like this :

xirsys1

To receive the api one need to make a one time call to their service , the result of which contains the keys to invoke the turn services from webrtc script .


&lt;script src=&quot;http://ajax.googleapis.com/ajax/libs/jquery/2.1.0/jquery.min.js&quot;&gt;&lt;/script&gt;&lt;script&gt;// &lt;![CDATA[

$.post(&quot;https://api.xirsys.com/getIceServers&quot;, {
ident: &quot;altanai&quot;,
secret: &quot;&lt; your secret key &gt;&quot;,
domain: &quot; &lt; your doemain &gt;&quot;,
application: &quot;default&quot;,
room: &quot;default&quot;,
secure: 1
},
function(data, status) {
alert(&quot;Data: &quot; + data + &quot;n Status: &quot; + status);
console.log(&quot;Data: &quot; + data + &quot;nnStatus: &quot; + status);
});
});
&lt;/script&gt;

The resulting output should look like ( my keys are hidden with a red rectangle ofcourse )

xirsysedited

The process of adding a TURN / STUN to your webrtc script in JS is as follows :

iceServers:[
{“url”:”stun:turn2.xirsys.com”},
{“username”:”< put your API username>”,”url”:”turn:turn2.xirsys.com:443?transport=udp”,”credential”:”< put your API credentail>”},
{“username”:”< put your API username>”,”url”:”turn:turn2.xirsys.com:443?transport=tcp”,”credential”:”< put your API credentail>”}]

website : http://xirsys.com/technology/

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Posted in STUN and TURN

Tagged , , , , , , , ,

NAT traversal using STUN and TURN

Posted on March 11, 2015 | 1 comment

We know that WebRTC is web based real-time communications on browser-based platform using the browser’s media application programming interface (API) and adding our JavaScript & HTML5 t control the media flow .
WebRTC has enabled developers to build apps/ sites / widgets / plugins capable of delivering simultaneous voice/video/data/screen-sharing capability in a peer to peer fashion.

But something which escapes our attention is the way in which media ia traversing across the network. Ofcourse the webrtc call runs very smoothly when both the peers are on open public internet without any restrictions or firewall blocks . But the real problem begins when one of the peer is behind a Corporate/Enterprise network or using a different Internet service provider with some security restrictions . In such a case the normal ICE capability of WebRTC is not enough , what is required is a NAT traversal mechanism .

STUN and TURN server protocols handle session initiations with handshakes between peers in different network environments . In case of a firewall blocking a STUN peer-to-peer connection, the system fallback to a TURN server which provides the necessary traversing mechanism through the NAT.

Lets study from the start ie ICE . What is it and why is it used ?

ICE (Interactive Connectivity Establishment )  framework ( mandatory by WebRTC standards  ) find network interfaces and ports in Offer / Answer Model to exchange network based information with participating communication clients. ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN)

ICE is defined by RFC 5245 – Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols.

Sample WebRTC offer holding ICE candidates :

type: offer, sdp: v=0
o=- 3475901263113717000 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video data
a=msid-semantic: WMS dZdZMFQRNtY3unof7lTZBInzcRRylLakxtvc
m=audio 9 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:/v5dQj/qdvKXthQ2
a=ice-pwd:CvSEjVc1z6cMnhjrLlcbIxWK
a=ice-options:google-ice
a=fingerprint:sha-256 F1:A8:2E:71:4B:4E:FF:08:0F:18:13:1C:86:7B:FE:BA:BD:67:CF:B1:7F:19:87:33:6E:10:5C:17:42:0A:6C:15
a=setup:actpass
a=mid:audio
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
m=video 9 RTP/SAVPF 100 116 117 96
c=IN IP4 0.0.0.0
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:/v5dQj/qdvKXthQ2
a=ice-pwd:CvSEjVc1z6cMnhjrLlcbIxWK
a=ice-options:google-ice
a=fingerprint:sha-256 F1:A8:2E:71:4B:4E:FF:08:0F:18:13:1C:86:7B:FE:BA:BD:67:CF:B1:7F:19:87:33:6E:10:5C:17:42:0A:6C:15
a=setup:actpass
a=mid:video
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
m=application 9 DTLS/SCTP 5000
c=IN IP4 0.0.0.0
a=ice-ufrag:/v5dQj/qdvKXthQ2
a=ice-pwd:CvSEjVc1z6cMnhjrLlcbIxWK
a=ice-options:google-ice
a=fingerprint:sha-256 F1:A8:2E:71:4B:4E:FF:08:0F:18:13:1C:86:7B:FE:BA:BD:67:CF:B1:7F:19:87:33:6E:10:5C:17:42:0A:6C:15
a=setup:actpass
a=mid:data
a=sctpmap:5000 webrtc-datachannel 1024

Notice the ICE candidates under video and audio . Now take a look at the SDP answer

type: answer, sdp: v=0
o=- 6931590438150302967 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video data
a=msid-semantic: WMS R98sfBPNQwC20y9HsDBt4to1hTFeP6S0UnsX
m=audio 1 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:WM/FjMA1ClvNb8xm
a=ice-pwd:8yy1+7x0PoHZCSX2aOVZs2Oq
a=fingerprint:sha-256 7B:9A:A7:43:EC:17:BD:9B:49:E4:23:92:8E:48:E4:8C:9A:BE:85:D4:1D:D7:8B:0E:60:C2:AE:67:77:1D:62:70
a=setup:active
a=mid:audio
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
m=video 1 RTP/SAVPF 100 116 117 96
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:WM/FjMA1ClvNb8xm
a=ice-pwd:8yy1+7x0PoHZCSX2aOVZs2Oq
a=fingerprint:sha-256 7B:9A:A7:43:EC:17:BD:9B:49:E4:23:92:8E:48:E4:8C:9A:BE:85:D4:1D:D7:8B:0E:60:C2:AE:67:77:1D:62:70
a=setup:active
a=mid:video
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
m=application 1 DTLS/SCTP 5000
c=IN IP4 0.0.0.0
b=AS:30
a=ice-ufrag:WM/FjMA1ClvNb8xm
a=ice-pwd:8yy1+7x0PoHZCSX2aOVZs2Oq
a=fingerprint:sha-256 7B:9A:A7:43:EC:17:BD:9B:49:E4:23:92:8E:48:E4:8C:9A:BE:85:D4:1D:D7:8B:0E:60:C2:AE:67:77:1D:62:70
a=setup:active
a=mid:data
a=sctpmap:5000 webrtc-datachannel 1024

Call Flow for ICE

STUN call flow for WebRTC Offer Answer

STUN call flow for WebRTC Offer Answer

WebRTC needs SDP Offer to be send to the clientB Javascript code from clientA Javascript code . Client B uses this SDP offer to generate an SDP Answer for client A. The SDP ( as seen on chrome://webrtc-internals/ ) includes ICE candidates which punchs open ports in the firewalls.
However incase both sides are symmetric NATs the media flow gets blocked. For such a case TURN is used which tries to give a public ip and port mapped to internal ip and port so as to provide an alternative routing mechanism like a packet-mirror. It can open a DTLS connection and use it to key the SRTP-DTLS media streams, and to send DataChannels over DTLS.

In order to Understand this better consider various scenarios

1 . No Firewall present on either peer . Both connected to open public internet .

Diagrammatic representation of  this shown as follows :

WebRTC signalling and media flow on Open public network

WebRTC signalling and media flow on Open public network

In this case there is no restriction to signal or media flow and the call takes places smoothly in p2p fashion.

2.  Either one or both the peer ( could be many in case of multi conf call ) are present behind a firewall  or  restrictive connection or router configured for intranet .

In such a case the signal may pass with the use of default ICE candidates or simple ppensource google Stun server such as

iceServers:[
{ ‘url’: “stun:stun.l.google.com:19302”}]

Diagram :

WebRTC signalling when peers are behind firewalls

WebRTC signalling when peers are behind firewalls

However the media is restricted resulting in a black / empty / no video situation for both peers  . To combat such situation a relay mechanism such as TURN is required which essentially maps public ip to private ips thus creating a alternative route for media and data to flow through .

WebRTC media flow when peers are behind NAT . Uses TURN relay mechanism

WebRTC media flow when peers are behind NAT . Uses TURN relay mechanism

Peer config should look like :

var configuration =  {
iceServers: [
{ “url’:”stun::”},
{ “url”:”turn::”}
]
};

var pc = new RTCPeerConnection(configuration);

3. When the TURN server is also behind a firewall .  The config file of the turn server need to be altered to map the public and private IP

The diagrammatic description of this is as follows :

WebRTC media flow when peers are behind NAT and TURN server is behind NAT as well . TURN config files bind a public interface to private interface address.

WebRTC media flow when peers are behind NAT and TURN server is behind NAT as well . TURN config files bind a public interface to private interface address .

1 Comment

Posted in STUN and TURN

Tagged , , , , , ,

continue : Streaming / broadcasting Live Video call to non webrtc supported browsers and media players

Posted on February 26, 2015 | 6 comments

This blog is in continuation to the attempts / outcomes and problems in building a WebRTC to RTP media framework that successfully stream / broadcast WebRTC content to non webrtc supported browsers ( safari / IE ) / media players ( VLC )

Attempt 4: Stream the content to a WebRTC endpoint which is hidden in a video call . Pick the stream from vp8 object URL send to a streaming server

This process involved the following components :

  • WebRTC API : simplewebrtc on Chrome
  • Transfer mechanism from client to Streaming server:  webrtc media channel

Problems : No streaming server is qualified to handle a direct webrtc input and stream it on network .

Attempt 4.1 : Stream the content to a WebRTC endpoint . Do WebRTC Endpoint to RTP Endpoint bridge using Kurento APIs. 

Use the RTP port and ip address to input into a ffmpeg or gstreamer or VLC terminal command and out put a live H264 stream on another ip and port address .  

This process involved the following components :

  • API : Kurento
  • Transfer mechanism : HTML5 webrtc client -> application server hosting java -> media server -> application for webrtc media to RTP media conversation -> RTP player

Screenshots of attempts with Wowza to stream from a ip and port

kurentowowoza

problems :

  • The stream was black ie no video content .

Attempt 4.2 : Build a WebRTC Endpoint to Http endpoint in kurento and force the video audio encoding to be that of H264 and PCMU.

code for adding constraints to output media and forcing choice of codecs

MediaPipeline pipeline = kurento.createMediaPipeline();
    WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build();
    HttpGetEndpoint httpEndpoint=new HttpGetEndpoint.Builder(pipeline).build();

    org.kurento.client.Fraction fr= new org.kurento.client.Fraction(1, 30);         
    VideoCaps vc= new VideoCaps(VideoCodec.H264,fr);
    httpEndpoint.setVideoFormat(vc);

    AudioCaps ac= new AudioCaps(AudioCodec.PCMU, 65536);
    httpEndpoint.setAudioFormat(ac);

    webRtcEndpoint.connect(httpEndpoint);

code for using gstreamer filter to force the output in raw format . It is a alternate solution to above

//basic media operation of 1 pipeline and 2 endpoinst
MediaPipeline pipeline = kurento.createMediaPipeline();
WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build();
RtpEndpoint rtpEndpoint = new RtpEndpoint.Builder(pipeline).build();

//adding Gstream filters 
GStreamerFilter filter1 = new GStreamerFilter.Builder(pipeline, &quot;videorate max-rate=30&quot;).withFilterType(FilterType.VIDEO).build();
GStreamerFilter filter2 = new GStreamerFilter.Builder(pipeline, &quot;capsfilter caps=video/x-h264,width=1280,height=720,framerate=30/1&quot;).withFilterType(FilterType.VIDEO).build();
GStreamerFilter filter3 = new GStreamerFilter.Builder(pipeline, &quot;capsfilter caps=audio/x-mpeg,layer=3,rate=48000&quot;).withFilterType(FilterType.AUDIO).build();

//connecting all poin ts to one another 
webRtcEndpoint.connect (filter1); 
filter1.connect (filter2); 
filter2.connect (filter3); 
filter3.connect (rtpEndpoint);

// RTP SDP offer and answer
String requestRTPsdp = rtpEndpoint.generateOffer();
rtpEndpoint.processAnswer(requestRTPsdp);

problem : The output is still webm

Attempt 5  : Use a RTP SDP Endpoint ( ie a SDP file valid for a given session ) and use it to play the WebRTC media over Wowza streaming server

This process involved the following components

  1. WebRTC Stream and object URL of the blob containing VP8 media
  2. Kurento  WebRTC Endpoint  bridge to generate SDP
  3. Wowza Streaming server

code for kurento to generate a SDP file from WebRTC to RTP bridge

@RequestMapping(value = &quot;/rtpsdp&quot;, method = RequestMethod.POST)
private String processRequestrtpsdp(@RequestBody String sdpOffer)
throws IOException, URISyntaxException, InterruptedException {

//basic media operation of 1 pipeline and 2 endpoinst
MediaPipeline pipeline = kurento.createMediaPipeline();
WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build();
RtpEndpoint rtpEndpoint = new RtpEndpoint.Builder(pipeline).build();

//connecting all poin ts to one another 
webRtcEndpoint.connect (rtpEndpoint);

// RTP SDP offer and answer
String requestRTPsdp = rtpEndpoint.generateOffer();
rtpEndpoint.processAnswer(requestRTPsdp);

// write the SDP conector to an external file
PrintWriter out = new PrintWriter(&quot;/tmp/test.sdp&quot;);
out.println(requestRTPsdp);
out.close();

HttpGetEndpoint httpEndpoint = new HttpGetEndpoint.Builder(pipeline).build();
PlayerEndpoint player = new PlayerEndpoint.Builder(pipeline, requestRTPsdp).build();
httpEndpoint.connect(rtpEndpoint);
player.connect(httpEndpoint);

// Playing media and opening the default desktop browser
player.play();
String videoUrl = httpEndpoint.getUrl();
System.out.println(&quot; ------- video URL -------------&quot;+ videoUrl);

// send the response to front client
String responseSdp = webRtcEndpoint.processOffer(sdpOffer);

return responseSdp;
}

problems : wowza doesnt not recognize the WebRTC SDP and play the video

screenshot of wowza with SDP input

Screenshot from 2015-01-30 15:28:59

Attempt 5.1 : Use a RTP SDP Endpoint ( ie a SDP file valid for a given session ) and use it to play the WebRTC media over Default Ubuntu media player 

SDP file formed contains contents such as :

v=0
o=- 3631611195 3631611195 IN IP4 192.168.0.119
s=Kurento Media Server
c=IN IP4 192.168.0.119
t=0 0
m=audio 42802 RTP/AVP 98 99 0
a=rtpmap:98 OPUS/48000/2
a=rtpmap:99 AMR/8000/1
a=rtpmap:0 PCMU/8000
a=ssrc:2713728673 cname:user59375791@host-ad1117df
m=video 35946 RTP/AVP 96 97 100 101
a=rtpmap:96 H263-1998/90000
a=rtpmap:97 VP8/90000
a=rtpmap:100 MP4V-ES/90000
a=rtpmap:101 H264/90000
a=ssrc:93449274 cname:user59375791@host-ad1117df

problem : deformed media

screenshot of playing from a SDP file

Screenshot from 2015-01-29 17:42:21

Attempt 5.2 : Use a RTP SDP Endpoint ( ie a SDP file valid for a given session ) and use it to play the WebRTC media over VLC using socket input

problem : nothing plays

screenshot of VLC connected to play from socket and failure to play anything

Screenshot from 2015-01-21 17:49:52

Attempt 5.3: Create a WebRTC endpoint and connected it to RTP endpoint via media pipelines . Also make the RTP SDP offer and answering the same . Play with ffnpeg / ffplay / gst playbin

String requestRTPsdp = rtpEndpoint.generateOffer();
rtpEndpoint.processAnswer(requestRTPsdp);

Write the requestRTPsdp to a file and obtain a RTP connector endpoint with Application/SDP .It plays okay with gst playbin ( 10 secs without audio )

Successful attempt to play from a gst playbin

gst-launch -vvv playbin uri=file:///tmp/test.sdp 

donekurento streaming

but refuses to be played by VLC , ffplay and even wowza . The error generated with

ffmpeg -i test.sdp -vcodec copy -acodec copy -f mpegts output-file.ts

or

ffmpeg -re -i test.sdp -vcodec h264 -acodec mp3 -f mpegts “udp://192.168.4.26:5000”

are

Could not find codec parameter for stream1 ( video:h263, none ) .Other errors types are , Could not write header for output file output file is empty nothing was encoded

Error screenshots of trying to play the RTP SDP file with ffmpeg

ffmpeg error kurebto1 ffmpeg error kurebto2

Attempt 6 : Use a WebRTC capable media and streaming server ( eg Kurento )  to pick a live stream of VP8 . Convert the VP8 to H268  ( ffmpeg / RTP endpoint ) . Convert H268 to Mp4 using MP4 parser and pass to a streaming server  ( wowza)

In process . to be updated .

6 Comments

Posted in webrtc Media Stack

Tagged , , , , , , , ,

Streaming / broadcasting Live Video call to non webrtc supported browsers and media players

Posted on February 17, 2015 | 2 comments

As the title of this article suggests I am going to pen my attempts of streaming / broadcasting Live Video WebRTC call to non WebRTC supported browsers and media players such as VLC , ffplay , default video player in Linux etc .

I am currently attempting to do this by making my own MP4 engine from WebRTC feed . However I am sharing my past experiments in hope of helping someone whose objective is not the same as mine and might get some help from these threads .

Attempt 1 : use one to many brodcasting API :

<!DOCTYPE html>
<html id=”home” lang=”en”>

<head>
<meta http-equiv=”Content-Type” content=”text/html; charset=UTF-8″>
<meta charset=utf-8>
<meta name=”viewport” content=”width=device-width, initial-scale=1.0, user-scalable=no”>
<meta name=”author” content=”altanai”>
<meta http-equiv=”X-UA-Compatible” content=”IE=edge,chrome=1″>

<link rel=”stylesheet” type=”text/css” href=”style.css”>

</head>

<body>

<table class=”visible”>
<tr>
<td style=”text-align: right;”>
<input type=”text” id=”conference-name” placeholder=”Broadcast Name”>
</td>
<td>
<select id=”broadcasting-option”>
<option>Audio + Video</option>
<option>Only Audio</option>
<option>Screen</option>
</select>
</td>
<td>
<button id=”start-conferencing”>Start Broadcasting</button>
</td>
</tr>
</table>
<table id=”rooms-list” class=”visible”></table>

<div id=”participants”></div>

<script src=”RTCPeerConnection-v1.5.js”></script>
<script src=”firebase.js”></script>
<script src=”broadcast.js”></script>
<script src=”broadcast-ui.js”></script>

</body>

</html>
 

It uses API fromwebrtc-experiment.com. The broadcast is in one direction only where the viewrs are never asked for their mic / webcam permission .

problem : The broadcast is for WebRTC browsers only and doesnt support non webrtc players / browsers

Attempt 1.1: Stream the media directly to nodejs through websocket


window.addEventListener('DOMContentLoaded', function() {

var v = document.getElementById('v');
navigator.getUserMedia = (navigator.getUserMedia || 
navigator.webkitGetUserMedia || 
navigator.mozGetUserMedia || 
navigator.msGetUserMedia);

if (navigator.getUserMedia) {
// Request access to video only
navigator.getUserMedia(
{
video:true,
audio:false
}, 
function(stream) {
var url = window.URL || window.webkitURL;
v.src = url ? url.createObjectURL(stream) : stream;
v.play();

var ws = new WebSocket('ws://localhost:3000', 'echo-protocol');
waitForSocketConnection(ws, function(){

console.log(" url.createObjectURL(stream)-----", url.createObjectURL(stream))
ws.send(stream);

console.log("message sent!!!"); 
});

},
function(error) {
alert('Something went wrong. (error code ' + error.code + ')');
return;
}
);
}
else {
alert('Sorry, the browser you are using doesn\'t support getUserMedia');
return;
}
});

//Make the function wait until the connection is made...
function waitForSocketConnection(socket, callback){
setTimeout(
function () {
if (socket.readyState === 1) {
console.log("Connection is made")
if(callback != null){
callback();
}
return;

} else {
console.log("wait for connection...")
waitForSocketConnection(socket, callback);
}

}, 5); // wait 5 milisecond for the connection...
}

problem : The video is in form of buffer and doesnot play

Attempt 2: Record the WebRTC media ( 5 secs each ) into chunks of webm format->  transfer them to other end -> append the chunks together like a regular file 

This process involved the following components :

  • Recorder Javascript library : RecordJs
  • Transfer mechanism : Record using RecordRTC.js -> send to other end for media server -> stitching together the small webm files into big one at runtime and play
  • Programs :

Code for video recorder

navigator.getUserMedia(videoConstraints, function(stream) {

video.onloadedmetadata = function() {
video.width = 320;
video.height = 240;

var options = {
type: isRecordVideo ? 'video' : 'gif',
video: video,
canvas: {
width: canvasWidth_input.value,
height: canvasHeight_input.value
}
};

recorder = window.RecordRTC(stream, options);
recorder.startRecording();
};
video.src = URL.createObjectURL(stream);
}, function() {
if (document.getElementById('record-screen').checked) {
if (location.protocol === 'http:')
alert('&lt;https&gt; is mandatory to capture screen.');
else
alert('Multi-capturing of screen is not allowed. Capturing process is denied. Are you enabled flag: "Enable screen capture support in getUserMedia"?');
} else
alert('Webcam access is denied.');
});

Code for video append-er

var FILE1 = '1.webm';
var FILE2 = '2.webm';
var FILE3 = '3.webm';
var FILE4 = '4.webm';
var FILE5 = '5.webm';

var NUM_CHUNKS = 5;
var video = document.querySelector('video');

window.MediaSource = window.MediaSource || window.WebKitMediaSource;
if (!!!window.MediaSource) {
alert('MediaSource API is not available');
}

var mediaSource = new MediaSource();

video.src = window.URL.createObjectURL(mediaSource);

function callback(e) {

var sourceBuffer = mediaSource.addSourceBuffer('video/webm; codecs="vorbis,vp8"');

GET(FILE1, function(uInt8Array) {

var file = new Blob([uInt8Array], {type: 'video/webm'});
var i = 1;

(function readChunk_(i) {

var reader = new FileReader();

reader.onload = function(e) {

sourceBuffer.appendBuffer(new Uint8Array(e.target.result));

if (i == NUM_CHUNKS) mediaSource.endOfStream();

else {
if (video.paused) {
video.play(); // Start playing after 1st chunk is appended.
}
readChunk_(++i);
}

};

reader.readAsArrayBuffer(file);

})(i); // Start the recursive call by self calling.
});
}

mediaSource.addEventListener('sourceopen', callback, false);
mediaSource.addEventListener('webkitsourceopen', callback, false);
mediaSource.addEventListener('webkitsourceended', function(e) {
logger.log('mediaSource readyState: ' + this.readyState);
}, false);

// function get the video via XHR
function GET(url, callback) {

var xhr = new XMLHttpRequest();
xhr.open('GET', url, true);
xhr.responseType = 'arraybuffer';
xhr.send();

xhr.onload = function(e) {

if (xhr.status != 200) {
alert("Unexpected status code " + xhr.status + " for " + url);
return false;
}

callback(new Uint8Array(xhr.response));
};
}

Shortcoming of this approach

  1. The webm files failed to play on most of the media players
  2. The recorder can only either record video or audio file at a time .

Attempt 2.1: Record the WebRTC media ( 5 secs each ) into chunks of webm format ( RecordRTC.js) >  Use Kurento JS script ( kws-media-api,js) to make a HTTP Endpoint to recorded Webm files  -> append the chunks together like a regular file at runtime 


function getByID(id) {
return document.getElementById(id);
}

var recordAudio = getByID('record-audio'),
recordVideo = getByID('record-video'),
stopRecordingAudio = getByID('stop-recording-audio'),
stopRecordingVideo = getByID('stop-recording-video'),
broadcasting=getByID('broadcasting');

var canvasWidth_input = getByID('canvas-width-input'),
canvasHeight_input = getByID('canvas-height-input');

var video = getByID('video');
var audio = getByID('audio');

var videoConstraints = {
audio: false,
video: {
mandatory: {},
optional: []
}
};

var audioConstraints = {
audio: true,
video: false
};

const ws_uri = 'ws://localhost:8888/kurento';
var URL_SMALL="http://localhost:8080/streamtomp4/approach1/5561840332.webm";


var audioStream;
var recorder;

recordAudio.onclick = function() {
if (!audioStream)
navigator.getUserMedia(audioConstraints, function(stream) {

if (window.IsChrome) stream = new window.MediaStream(stream.getAudioTracks());
audioStream = stream;

audio.src = URL.createObjectURL(audioStream);
audio.muted = true;
audio.play();

// "audio" is a default type
recorder = window.RecordRTC(stream, {
type: 'audio'
});
recorder.startRecording();
}, function() {});
else {
audio.src = URL.createObjectURL(audioStream);
audio.muted = true;
audio.play();
if (recorder) recorder.startRecording();
}


window.isAudio = true;

this.disabled = true;
stopRecordingAudio.disabled = false;
};

stopRecordingAudio.onclick = function() {
this.disabled = true;
recordAudio.disabled = false;
audio.src = '';

if (recorder)
recorder.stopRecording(function(url) {
audio.src = url;
audio.muted = false;
audio.play();

document.getElementById('audio-url-preview').innerHTML = '&lt;a href="' + url + '" target="_blank"&gt;Recorded Audio URL&lt;/a&gt;';
});
};

recordVideo.onclick = function() {
recordVideoOrGIF(true);
};


function recordVideoOrGIF(isRecordVideo) {
navigator.getUserMedia(videoConstraints, function(stream) {

video.onloadedmetadata = function() {
video.width = 320;
video.height = 240;

var options = {
type: isRecordVideo ? 'video' : 'gif',
video: video,
canvas: {
width: canvasWidth_input.value,
height: canvasHeight_input.value
}
};

recorder = window.RecordRTC(stream, options);
recorder.startRecording();
};
video.src = URL.createObjectURL(stream);
}, function() {
if (document.getElementById('record-screen').checked) {
if (location.protocol === 'http:')
alert('&lt;https&gt; is mandatory to capture screen.');
else
alert('Multi-capturing of screen is not allowed. Capturing process is denied. Are you enabled flag: "Enable screen capture support in getUserMedia"?');
} else
alert('Webcam access is denied.');
});

window.isAudio = false;

if (isRecordVideo) {
recordVideo.disabled = true;
stopRecordingVideo.disabled = false;
} else {
recordGIF.disabled = true;
stopRecordingGIF.disabled = false;
}
}

stopRecordingVideo.onclick = function() {
this.disabled = true;
recordVideo.disabled = false;

if (recorder)
recorder.stopRecording(function(url) {
video.src = url;
video.play();
document.getElementById('video-url-preview').innerHTML = '&lt;a href="' + url + '" target="_blank"&gt;Recorded Video URL&lt;/a&gt;';

});
};


/*--------------------------broadcasting -----------------------------------*/

function onerror(error)
{
console.log( " error occured");
console.error(error);
};

broadcast.onclick = function() {
var videoOutput = document.getElementById("videoOutput");

KwsMedia(ws_uri, function(error, kwsMedia)
{
if(error) return onerror(error);

// Create pipeline
kwsMedia.create('MediaPipeline', function(error, pipeline)
{
if(error) return onerror(error);

// Create pipeline media elements (endpoints &amp; filters)
pipeline.create('PlayerEndpoint', {uri: URL_SMALL},
function(error, player)
{
if(error) return console.error(error);

pipeline.create('HttpGetEndpoint', function(error, httpGet)
{
if(error) return onerror(error);

// Connect media element between them
player.connect(httpGet, function(error, pipeline)
{
if(error) return onerror(error);
// Set the video on the video tag
httpGet.getUrl(function(error, url)
{
if(error) return onerror(error);

videoOutput.src = url;

console.log(url);

// Start player
player.play(function(error)
{
if(error) return onerror(error);

console.log('player.play');
});
});
});

// Subscribe to HttpGetEndpoint EOS event
httpGet.on('EndOfStream', function(event)
{
console.log("EndOfStream event:", event);
});
});
});
});
},
onerror);

}

problem : dissecting the live video into small the files and appending to each other on reception is an expensive , time and resource consuming process . Also involves heavy buffering and other problems pertaining to real-time streaming .

Attempt 2.2 : Send the recorded chunks of webm to a port on linux server . Use socket programming to pick up these individual files and play using  VLC player from UDP port of the Linux Server

Screenshot from 2015-01-22 15:32:51

Attempt 2.3: Send the recorded chunks of webm to a port on linux server socket . Use socket programming to pick up these individual webm files and convert to H264 format so that they can be send to a media server. 

This process involved the following components :

  • Recorder Javascript library : RecordJs
  • Transfer mechanism :WebRTC endpoint -> Call handler ( Record in chunks ) -> ffmpeg / gstreamer to put it on RTP -> streaming server like wowza – > viewers
  • Programs : Use HTML webpage Webscoket connection -> nodejs program to write content from websocket to linux socket -> nodejs program to read that socket and print the content on console

Program to transfer the webm recorder files over websocket to nodejs program

//Make the function wait until the connection is made...
function waitForSocketConnection(socket, callback){
setTimeout(
function () {
if (socket.readyState === 1) {
console.log("Connection is made")
if(callback != null){
callback();
}
return;

} else {
console.log("wait for connection...")
waitForSocketConnection(socket, callback);
}

}, 5); // wait 5 milisecond for the connection...
}

function previewFile() {
var preview = document.querySelector('img');
var file = document.querySelector('input[type=file]').files[0];
var reader = new FileReader();

reader.onloadend = function () {

preview.src = reader.result;
console.log(" reader result ", reader.result);

var video=document.getElementById("v");
video.src=reader.result;
console.log(" video played ");

var ws = new WebSocket('ws://localhost:3000', 'echo-protocol');

waitForSocketConnection(ws, function(){
ws.send(reader.result); 
console.log("message sent!!!"); 
});

}

if (file) {
// converts to base64 encoded string of the file data
//reader.readAsDataURL(file);

reader.readAsBinaryString(file);

} else {
preview.src = "";
}
}

Program for Linux Sockets sender which creates the socket for the webm files

var net = require('net');
var fs = require('fs');
var socketPath = '/tmp/tfxsocket';
var http = require('http');
var stream = require('stream');
var util = require('util');

var WebSocketServer = require('ws').Server;
var port = 3000;
var serverUrl = "localhost";

var socket;
/*--------------------------------http server -----------------------------*/
var server= http.createServer(function (request, response) {

});

server.listen(port, serverUrl);

console.log('HTTP Server running at ',serverUrl,port);

/*--------------------------------websocket server -----------------------------*/

var wss = new WebSocketServer({server: server});

wss.on("connection", function(ws) {
console.log("websocket connection open");

ws.on('message', function (message) {
console.log(" stream recived from broadcast client on port 3000 ");

var s = require('net').Socket();
s.connect(socketPath);
s.write(message);

console.log(" send the stream to socketPath",socketPath); 
});

ws.on("close", function() {
console.log("websocket connection close")
});

});

Program for Linux Socket Listener using nodejs and socket . Here the socket is in node /tmp/mysocket

var net = require('net');

var client = net.createConnection("/tmp/mysocket");

client.on("connect", function() {
console.log("connected to mysocket");
});

client.on("data", function(data) {
console.log(data);
});

client.on('end', function() {
console.log('server disconnected');
});

Output 1: Video Buffer displayed

Screenshot from 2015-01-22 15:35:06 (copy)

Output 2 : Random data from Video displayed

Screenshot from 2015-01-23 12:57:35

ffmpeg format of transfering the content from socket to UDP IP and port

ffmpeg -i unix://tmp/mysocket -f format udp://192.168.0.119:8083

problems of this approach : The video was on a passing stage from the socket and contained no information as such when tried to play / show console

Attempt 3 : Send the live WebRTC stream from Kurento WebRTC endpoint to Kurento HTTP endpoint . play using  Mozilla VLC web plugin

VLC mozilla plugin can be embedded by :


name=”video2″
autoplay=”yes” loop=”no” hidden=”no”
target=”rtp://@192.165.0.119:8086″ />

screenshot of failure on part of Mozilla VLC plugin to play from a WebRTC endpoint

Screenshot from 2015-01-29 10:37:06Screenshot from 2015-01-29 10:37:17

Screenshot from 2015-01-29 12:06:14

problem : VLC mozilla plugin was unable to play the video

………………………………………………………………………………………………………………..

The 4th , 5th and 6th sections of this article are in the next blog :

continue : Streaming / broadcasting Live Video call to non webrtc supported browsers and media players

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Posted in webrtc Media Stack

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TFX platform

Posted on January 17, 2015 | Leave a comment

So I haven’t written anything worthy in a while , just published some posts that were lying around in my drafts . Here I write about the main thing . some thing awesome that I was trying to accomplish in the last quarter .

<< TFX is now live in chrome store , open and free for public use . No signin or account required , no advertisements   : https://chrome.google.com/webstore/detail/tfx-sessions/aochimdcllmgleokpnlabijehdlmkdga >>

TFX Sessions is a plug and play platform for VoIP ( voice over IP ) scenarios.  Intrinsically it  is a very lightweight API package and shipped in form of a Chrome Extension . It is a turn-key solution when parties want instant audio/audio communication without any sign-in ,plugin installation or additional downloads  . Additionally TFX Sessions is packaged with some interesting plugins which enable the communicating parties to get the interactive and immersive experience as in a face to face meeting.

There is a market requirement of making a utterly simple WebRTC API  that has everything needed to build bigger aggregate projects but the available solutions are either just to basic or much too complex . So I initially started writing my own getuserMedia APIs, but left it midway and picked up simplewebrtc API instead for want of time .Then I focused on the main crux  of the project which was the widget API and ease of integration.

How Does TFX Sessions Work ?

  • Signalling channel establishes the session using Offer- Answer Model
  • Browser’s  media API’s , like getUserMedia and Peerconnection are used for media flow
  • Media only flows peer to peer

TFX WebRTC platform architecture . socket io signalling

TFX WebRTC platform architecture . socket io signalling

A Widget is essentially any web project that wants communication over webrtc channel . Once the platform is ready I have core APIs , widgets and signalling server. Then came up the subject of enterprise internet blocking my communication stream . Time for TURN ( Coturn in my case ).

TFX create / join room

TFX create / join room

TFX startup screen

TFX startup screen

Components of TFX

Client Side Components of tangoFX :

broplug API
Inhouse master library for TangoFX. Makes the TFX sessions platform .Masks the low level webrtc and socketio functions .
Provides simple to use handles for interesting plugins development in platform .
simplewebrtc
performs webrtc support , peer configuration , wildemitter , utils , event emitters , JSON formatter , websocket , socket namespace , transport , XHR more like so
socket.io.js
exports and listener for real time event based bidirectional communication
Jquerry
JS library for client side scripting
Bootstrap
HTML and CSS-based design templates for typography, forms, buttons, navigation and other interface components.

Server Side components

Signalling Server -signal master
socket.io server for webrtc signalling
TURN -Coturn
TURN protocol based media traversal for connecting media across restricting domains ie firewalls, network policies etc .
redis Data structure to maintain current and lost sessions

TFXSessions Components

Architecture

So here is the final architecture of TFX chrome extension widget based platform .

  • The client side contains widgets , chore extension APIs , chrome’s WebRTC API’s , socket.io client for signalling , HTML5 , Jquerry , Javascript , and CSS for styling.
  • The Server Side of the solution contains socket.io server for signalling , manuals and other help/support materials , HTTPS certificate and TURN server implementation for NAT .

TFX whitepaper v2.0

TFX platform Server client components . WebRTC media and socketio communication . Build as chrome Extension

Salient Features

  • The underlying technology of TangoFX is  webrtc with socket based signalling  . Also it adheres to the latest standards of W3C , IETF and ITU on internet telephony .
  • TangoFX sessions is extremely scalable and flexible due to the abstraction between communication and service development. This make it a piece of  cake for any web developer using  TangoFX interface to add his/ her own service easily and quickly without diving into the nitty gritties .
  • TangoFX is currently packaged in a chrome extension supported on chrome browser on desktop operating system like window , mac , linux etc .
  • The call is private to both the parties as it is peer-to-peer meaning that the media / information exchanged by the parties over TangoFX does not pass through an intervening server as in other existing internet calling solutions.
  • TangoFX is very adaptive to slow internet and can be used across all kinds of networks such as corporate to public without being affected by firewall or restricting policies  .

TFX Widget Screens

Alright so that’s there . Tada the platform is alive and kicking . Right now in beta stage however . Intensive testing going on here . However here are some screenshots that are from my own developer version .

TFX recording widget

TFX recording widget

TFX face detection and overlay widget

TFX face detection and overlay widget

TFX multilingual communication

TFX multilingual communication

TFX screen-sharing

TFX screen-sharing

TFX video Filters

TFX video Filters

TFX audio visualizer

TFX audio visualizer

TFX text messaging widget

TFX text messaging widget

TFX cross domian access . flicker here

TFX cross domain access . flicker here

TFX draw widget

TFX draw widget

TFX code widget supportes many programming languages

TFX code widget supportes many programming languages

TFX webrtc dynamic stats

TFX webrtc dynamic stats

TFX introduction widget

TFX introduction widget

Note that the widgets described above have been made with the help of third party APIs.

TFX Sessions Summary

We saw that TFX is WebRTC based communication and collaboration solution .It is build on Open standards from w3c , IETF , Google etc.
Scalable and customizable. Immersive and interactive experience .
Easy to build widgets framework using TangoFX APIs.

TFX User Manual : https://tfxserver.above-inc.com/static/manuals/material/TangoFXv0.1Usersmanual.pdf

TFX Developer’s Manual : https://tfxserver.above-inc.com/static/manuals/material/TangoFXv0.1Developersmanual.pdf

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Posted in WebRTC SaaS

Tagged , , , , ,

APPRTC , Talky.io , TokBox

Posted on December 2, 2014 | Leave a comment

In the last few months I have been observing how the course for WebRTC is turning out so far . Unfortunately contrary to my expectations the fundamental holes in WebRTC specification are still the same with less being done to fulfill them . Ofcourse now there are abundance of interactive
WebRTC API each using a new masking method to call the same old WebRTC API function of getusermedia and peer-connection . Few of these I will list down in this blog but no concrete stable reliable  guide to setup the backbone network ( yes i am referring to Media inter conversion , relay , TURN , STUN servers ) which is left to telecom software engineer / developer to find out and configure . Instead I see many commercial service providers who claim of providing their backend for our WebRTC implementation but that in my opinion completely defeats the objective of WebRTC based communication .  WebRTC was meant to *everything you can’t do with proprietary communication tools and networks* .

Well moving on , here are some nice API implementations of WebRTC ( only for Websockets no SIP )

1.appRTC

apprtc apprtc2 apprtc3 apprtc4 apprtc5

2. talky.io

talky3 talky2 talky1 talky4 talky5 talky63. tokbox

tokbox2tokbox1tokbox7 tokbox6 tokbox5 tokbox4 tokbox3

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Posted in webrtc APIs

Tagged , , , , ,

Call Continuity from Mobile GSM network to WebRTC

Posted on October 20, 2014 | 1 comment

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 number 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.

Find the diagram depicting this below :

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

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Posted in webrtc service

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