HTTP/2 – offer/answer signaling for WebRTC call

HTTP ( Hyper Text Transfer Protocol ) is the top application layer protocol atop the Tarnsport layer ( TCP ) and the Network layer ( IP ).


HTTP/1.1 was released in 1997. HTTP/1 allowed only 1 req at a time. But HTTP/1.1 allows one one outstanding connection on a TCP session but allowed request pieplinig to achieve concurency.


HTTP/2 was released in 2015, it aimed at reducing latency while delivering heavy graphics, videos and other media cpmponents on web page especially on mobile sites .
optimizes server push and service workers.


  • Header compression (HPACK)
  • reuse connection TCP connection
    • All frames (e.g. HEADERS, DATA, etc) are sent over single TCP connection
  • Binary framing layer
    • Prioritization
    • Flow control
    • Server push
  • Request → Stream
    • Streams multiplexed
    • Streams prioritized
  • (+) low latency / iproves end user perceived latency
  • (+) retain semantics of HTTP1.1


A key differenet between HTTP/1.1 and HTTP/2 is the fact that former transmites requests and reponses in plaintext whereas the later encapsulates them into binary format, proving more features and scope for optimzation. Thus at protocol level, it is all about frames of bytes which are part of stream.

  • HTTP messages are decomposed into one or more frames
    • HEADERS for meta-data (9-byte, length prefixed)
    • DATA for payload
    • RST_STREAM to cancel
    • …..

“enables a more efficient use of network resources and a reduced perception of latency by introducing header field compression and allowing multiple concurrent exchanges on the same connection. It also introduces unsolicited push of representations from servers to clients.”

Hypertext Transfer Protocol Version 2 (HTTP/2) draft-ietf-httpbis-http2-latest

It is important to know that Browsers only implement HTTP/2 under HTTPS, thus TLS connection is must for whichw e need certs ad keys signed by CA ( either self signed using openssl , signed by public CA like godaddy , verisign or letsencrypt).

Data Flow

DATA frames are subject to per-stream and connection flow control

  • Flow control allows the client to pause stream delivery, and resume it later

Compatibility Layer between HTTP1.1 and HTTP2.0 in node

Nodejs >9 provides http2 as native module. Exmaple of using http2 with compatibility layer

const http2 = require('http2');
const options = {
 key: 'ss/key', // path to key
 cert: 'ssl/cert' // path to cert

const server = http2.createSecureServer(options, (req, res) => {
    req.addListener('end', function () {
        file.serve(req, res);

in replacement for existing server http/https server

const https = require('https');
app = https.createServer(options, function (request, response) {
    request.addListener('end', function () {
        file.serve(request, response);

app.listen(8084); Websocket over HTTP2

The WebSocket Protocol uses the HTTP/1.1 Upgrade mechanism to transition a TCP connection from HTTP into a WebSocket connection

Due to its multiplexing nature, HTTP/2 does not allow connection-wide header fields or status codes, such as the Upgrade and Connection request-header fields or the 101 (Switching Protocols) response code. These are all required for opening handshake.

Ideally the code shouldve looekd like this with backward compatiability layer , but continue reading update ..

var app = http2
    .createSecureServer(options, (req, res) => {
        req.addListener('end', function () {
            file.serve(req, res);

var io = require('').listen(app);'*:*');
io.on('connection', onConnection); // evenet handler onconnection

Error during WebSocket handshake: Unexpected response code: 403

update May 2020 : I tried using the http2 server with websocket like mentioned above ,h owever many many hours of working around WSS over HTTP2 secure server , I consistencly kept faccing the ECONNRESET issues after couple of seconds , which would crash the server

client 403

Therefore leaving the web server to server htmll conetnt I reverted the siganlling back to HTTPs/1.1 given the reasons for sticking with WSS is low latency and existing work that was already put in.

Example Repo :

Reading Further of exploring HTTP CONNECT methods for setting WS handshake . Will update this section in future if it works .


A “stream” is an independent, bidirectional sequence of frames exchanged between the client and server within an HTTP/2 connection.
A single HTTP/2 connection can contain multiple concurrently open streams, with either endpoint interleaving frames from multiple streams.

Core http2 module provides new core API (Http2Stream), accessed via a “stream” listener:

const http2 = require('http2');
const options = {
 key: 'ss/key', // path to key
 cert: 'ssl/cert' // path to cert

const server = http2.createSecureServer(options, (stream, headers) => {
    stream.respond({ ':status': 200 });
    stream.end('some text!');

Other features

  • stream multiplexing
  • stream Prioritization
  • header compression
  • Flow Control
  • support for trailer

Persistent , one connection per origin.

With the new binary framing mechanism in place, HTTP/2 no longer needs multiple TCP connections to multiplex streams in parallel; each stream is split into many frames, which can be interleaved and prioritized. As a result, all HTTP/2 connections are persistent, and only one connection per origin is required,

Server Push

bundling multiple assets and resources into a single HTTP/2  and lets the srever proactively push resources to client’s cache .

Server issues PUSH_PROMISE , client validates whether it needs the resource of not. If the client matches it then they will load like regular GET call

The PUSH_PROMISE frame includes a header block that contains a complete set of request header fields that the server attributes to the request.

After sending the PUSH_PROMISE frame, the server can begin delivering the pushed response as a response on a server-initiated stream that uses the promised stream identifier.

Client receives a PUSH_PROMISE frame and can either chooses to accept the pushed response or if it does not wish to receive the pushed response from the server it can can send a RST_STREAM frame, using either the CANCEL or REFUSED_STREAM code and referencing the pushed stream’s identifier.

Push Stream Support


respondWithFile() and respondWithFD() APIs can send raw file data that bypasses the Streams API.

Related technologies


Multipurpose Internet Mail Extensions (MIME) is an Internet standard that extends the format of email messages to support text in character sets other than ASCII, as well as attachments of audio, video, images, and application programs. Message bodies may consist of multiple parts, and header information may be specified in non-ASCII character sets.

Email messages + MIME : transmitted with standard protocols, such as the Simple Mail Transfer Protocol (SMTP), the Post Office Protocol (POP), and the Internet Message Access Protocol (IMAP).

MIME in HTTP in WWW : servers insert a MIME header field at the beginning of any Web transmission. Clients use the content type or media type header to select an appropriate viewer application for the type of data indicated. Browsers typically contain GIF and JPEG image viewers.

MIME header fields

  • MIME version
MIME-Version: 1.0
  • Content Type
Content-Type: text/plain

multipart/mixed , text/html, image/jpeg, audio/mp3, video/mp4, and application/msword

  • content disposition
Content-Disposition: attachment; filename=genome.jpeg;
  modification-date="Wed, 12 Feb 1997 16:29:51 -0500";
  • Content-Transfer-Encoding


source :

References :

Websockets as VOIP signal transport medium

Web resources are usually build on request/response paradigm such as HTTP , SIP messages . This means that server responds only when a client requests it to. This made web intercations very slow and unsuited for VOIP signalling
Long Poll involved repeated polling checks to load new server resources by itself instead of client made explicit request
AJAX and multipart XHR tried to patch the problem by selective reloading however they still required that client perform the mapping for an incomig reply to map to correct request.
However due to overhead latency involved with HTTP transaction and its working mode to open new TCP connetion for every request and reponse and add HTTP headers, none of them were suited to realtime operations

Websocket is the current (2017) most idelistic solution to perform realtime sigalling suited to VOIP requirnments due to its nature os establish a socket .

Websocket Protocol

Enables two-way communication between a client running untrusted code in a controlled environment to a remote host that has opted-in to communications from that code.

protocol consists of an opening handshake followed by basic message framing, layered over TCP.
handshake is interpreted by HTTP servers as an Upgrade request.

Secure websocket example :

Request URL: wss://
Request Method: GET
Status Code: 101 Switching Protocols

Response Headers
Connection: Upgrade
Sec-WebSocket-Accept: UVhTdFOWfywGyQTKDRZyGuhkfls=
Sec-WebSocket-Extensions: permessage-deflate
Upgrade: websocket

Request Headers
Accept-Encoding: gzip, deflate, br
Accept-Language: en-US,en;q=0.9
Cache-Control: no-cache
Connection: Upgrade
Pragma: no-cache
Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits
Sec-WebSocket-Key: 06FNaHge8GLGVuPFxV2fAQ==
Sec-WebSocket-Version: 13
Upgrade: websocket
User-Agent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_14_6) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/76.0.3809.100 Safari/537.36

Query String parameters
transport: websocket
sid: hh3Dib_aBWgqyO1IAAEL

Working with websockets

A new websocket can be opned with ws or wss and it can have sub protocols like in example .

var wsconnection = new WebSocket('wss://', ['soap', 'xmpp']);

It can be attached with event handlers

wsconnection.onopen = function () {
wsconnection.onerror = function (error) {
  console.log('WebSocket Error ' + error);
wsconnection.onmessage = function (e) {
  console.log('message received : ' +;

Send Data on websocket

message string


Blob or ArrayBuffer object to send binary data
Ex : Sending canvas ImageData as ArrayBuffer

var img = canvas_context.getImageData(0, 0, 400, 320);
var binary = new Uint8Array(;
for (var i = 0; i <; i++) {
  binary[i] =[i];

Ex : sending file as Blob

var file = document.querySelector('input[type="file"]').files[0];

Closing the connection

if (socket.readyState === WebSocket.OPEN) {

Registry for Close codes for WS
1000 Normal Closure [IESG_HYBI] [RFC6455]
1001 Going Away [IESG_HYBI] [RFC6455]
1002 Protocol error [IESG_HYBI] [RFC6455]
1003 Unsupported Data [IESG_HYBI] [RFC6455]
1004 Reserved [IESG_HYBI] [RFC6455]
1005 No Status Rcvd [IESG_HYBI] [RFC6455]
1006 Abnormal Closure [IESG_HYBI] [RFC6455]
1007 Invalid frame payload data [IESG_HYBI] [RFC6455]
1008 Policy Violation [IESG_HYBI] [RFC6455]
1009 Message Too Big [IESG_HYBI] [RFC6455]
1010 Mandatory Ext. [IESG_HYBI] [RFC6455]
1011 Internal Error [IESG_HYBI] [RFC6455][RFC Errata 3227]
1012 Service Restart [Alexey_Melnikov] []
1013 Try Again Later [Alexey_Melnikov] []
1014 The server was acting as a gateway or proxy and received an invalid response from the upstream server. This is similar to 502 HTTP Status Code. [Alexey_Melnikov] []
1015 TLS handshake [IESG_HYBI] [RFC6455]
1016-3999 Unassigned
4000-4999 Reserved for Private Use [RFC6455]

WebSocket Subprotocol Name Registry

  • MBWS
  • soap soap
  • wamp WAMP (“The WebSocket Application Messaging Protocol”)
  • v10.stomp Name: STOMP 1.0 specification
  • v11.stomp Name: STOMP 1.1 specification
  • v12.stomp Name: STOMP 1.2 specification
  • ocpp1.2 OCPP 1.2 open charge alliance
  • ocpp1.5 OCPP 1.5 open charge alliance
  • ocpp1.6 OCPP 1.6 open charge alliance
  • ocpp2.0 OCPP 2.0 open charge alliance
  • ocpp2.0.1 OCPP 2.0.1
  • rfb RFB [RFC6143]
  • sip WebSocket Transport for SIP (Session Initiation Protocol) [RFC7118]
  • OMA RESTful Network API for Notification Channel
  • wpcp Web Process Control Protocol (WPCP)
  • amqp Advanced Message Queuing Protocol (AMQP) 1.0+
  • mqtt mqtt [MQTT Version 5.0]
  • jsflow jsFlow pubsub/queue protocol
  • rwpcp Reverse Web Process Control Protocol (RWPCP)
  • xmpp WebSocket Transport for the Extensible Messaging and Presence Protocol (XMPP) [RFC7395]
  • ship SHIP – Smart Home IP SHIP (Smart Home IP) is a an IP based approach to plug and play home automation and smart energy / energy efficiency, which can easily be extended to additional domains such as Ambient Assisted Living (AAL). SHIP can be used solely on the customer premises or can be integrated into a cloud based solution.
  • mielecloudconnect Miele Cloud Connect Protocol This protocol is used to securely connect household or professional appliances to an internet service portal via a public communication network in order to enable remote services.
  • Push Channel Protocol
  • msrp WebSocket Transport for MSRP (Message Session Relay Protocol) [RFC7977]
  • TLCP (Text Lightstreamer Client Protocol)
  • bfcp WebSocket Transport for BFCP (Binary Floor Control Protocol)
  • Softvelum Low Delay Protocol SLDP is a low latency live streaming protocol for delivering media from servers to MSE-based browsers and WebSocket-enabled applications.
  • opcua+uacp OPC UA Connection Protocol
  • opcua+uajson OPC UA JSON Encoding
  • v1.swindon-lattice+json Swindon Web Server Protocol (JSON encoding)
  • v1.usp USP (Broadband Forum User Services Platform)
  • mles-websocket mles-websocket
  • coap Constrained Application Protocol (CoAP) [RFC8323]
  • TLCP (Text Lightstreamer Client Protocol)
  • sqlnet This protocol is used for communication between Oracle database client and database server, and its usage as subprotocol of websocket is primarly geared towards cloud deployments. sqlnet supports bi-directional data transfer and is full duplex in nature.
  • oneM2M.R2.0.json oneM2M R2.0 JSON
  • oneM2M.R2.0.xml oneM2M R2.0 XML
  • oneM2M.R2.0.cbor oneM2M R2.0 CBOR
  • transit Transit
  • MPEG-DASH-ServerPush-23009-6-2017
  • MPEG-MMT-23008-1-2018
  • Softvelum WebSocket signaling protocol WebRTC live streaming requires WebSocket-based signaling protocol for every specific implementation. Softvelum products will use this subprotocol for signaling

websocket libraries

C++: libwebsockets
Java: Jetty
Node.JS: ws
PHP: Ratchet, phpws

Ref :
RFC 6455 – The websocket protocol
Websocket Protocol Registeries :
IANA websocket -


SIP stack written in C. Available under GPL

pjsip dev guide architecture diagram

PJSip user agent

local_info+tag, local_contact,

pj_status_t pjsip_ua_init(endpt, param);
pj_status_t pjsip_ua_destroy(void);
pjsip_module* pjsip_ua_instance(void);
pjsip_endpoint* pjsip_ua_get_endpt(ua);

PJSip dialog


state, session_counter, initial_cseq, local_cseq, remote_cseq, route_set,

local_info+tag, local_contact, remote_info+tag, remote_contact, next_set


pj_status_t pjsip_dlg_create_uac(ua, local_uri, contact, …);
pj_status_t pjsip_dlg_create_uas(ua, rdata, contact, &dlg);
pj_status_t pjsip_dlg_fork(old_dlg,rdata,&dlg);
pj_status_t pjsip_dlg_set_route_set(dlg, route_set);
pj_status_t pjsip_dlg_inc_session(dlg);
pj_status_t pjsip_dlg_dec_session(dlg);
pj_status_t pjsip_dlg_add_usage(dlg, mod);
pj_status_t pjsip_dlg_create_request(dlg,method,cseq,&tdata);
pj_status_t pjsip_dlg_send_request(dlg,tdata,&tsx);
pj_status_t pjsip_dlg_create_response(dlg,rdata,code,txt,&tdata);
pj_status_t pjsip_dlg_modify_response(dlg,tdata,code,txt);
pj_status_t pjsip_dlg_send_response(dlg,tsx,tdata);
pj_status_t pjsip_dlg_send_msg(dlg,tdata);
pjsip_dialog* pjsip_tsx_get_dlg(tsx);
pjsip_dialog* pjsip_rdata_get_dlg(rdata);

PJsip module

Attributes: name, id, priority, …

pj_bool_t on_rx_request(rdata);
pj_bool_t on_rx_response(rdata);
void on_tsx_state(tsx,event);

SDP state Offer/ Answer transition

SDP negotiator structure

SDP negotiator class diagram PJSIP dev guide


Pre requisities :

GNU make (other make will not work),
GNU binutils for the target, and
GNU gcc for the target.
In addition, the following libraries are optional, but they will be used if they are present:

ALSA header files/libraries (optional) if ALSA support is wanted.
OpenSSL header files/libraries (optional) if TLS support is wanted.

Video support

Video4Linux2 (v4l2) development library
​ffmpeg development library for video codecs: H.264 (together with libx264) and H263P/H263-1998.
Simple DirectMedia Layer SDL
libyuv for format conversion and video manipulation / Ffmpeg
OpenH264 / VideoToolbox (only for Mac) / ffmpeg

get tar and untar sourcecode

tar -xvzf pjproject-2.9.tar.bz2

configure to make ‘build.mak’, and ‘os-auto.mak’

./configure --enable-shared --disable-static --enable-memalign-hack --enable-gpl --enable-libx264

It runs checks , pay attention to traces such as


checking opus/opus.h usability... yes
checking opus/opus.h presence... yes
checking for opus/opus.h... yes
checking for opus_repacketizer_get_size in -lopus... yes
OPUS library found, OPUS support enabled

For more features customization `configure’ configures pjproject 2.x to adapt to many kinds of systems.

Usage: ./aconfigure [OPTION]… [VAR=VALUE]…

To assign environment variables (e.g., CC, CFLAGS…), specify them as
VAR=VALUE. See below for descriptions of some of the useful variables.

Defaults for the options are specified in brackets.

-h, –help display this help and exit
–help=short display options specific to this package
–help=recursive display the short help of all the included packages
-V, –version display version information and exit
-q, –quiet, –silent do not print checking ...' messages --cache-file=FILE cache test results in FILE [disabled] -C, --config-cache alias for–cache-file=config.cache’
-n, –no-create do not create output files
–srcdir=DIR find the sources in DIR [configure dir or `..’]

Installation directories:
–prefix=PREFIX install architecture-independent files in PREFIX
–exec-prefix=EPREFIX install architecture-dependent files in EPREFIX

By default, make install' will install all the files in /usr/local/bin’, /usr/local/lib' etc. You can specify an installation prefix other than/usr/local’ using --prefix', for instance–prefix=$HOME’.

For better control, use the options below.

Fine tuning of the installation directories:
–bindir=DIR user executables [EPREFIX/bin]
–sbindir=DIR system admin executables [EPREFIX/sbin]
–libexecdir=DIR program executables [EPREFIX/libexec]
–sysconfdir=DIR read-only single-machine data [PREFIX/etc]
–sharedstatedir=DIR modifiable architecture-independent data [PREFIX/com]
–localstatedir=DIR modifiable single-machine data [PREFIX/var]
–libdir=DIR object code libraries [EPREFIX/lib]
–includedir=DIR C header files [PREFIX/include]
–oldincludedir=DIR C header files for non-gcc [/usr/include]
–datarootdir=DIR read-only arch.-independent data root [PREFIX/share]
–datadir=DIR read-only architecture-independent data [DATAROOTDIR]
–infodir=DIR info documentation [DATAROOTDIR/info]
–localedir=DIR locale-dependent data [DATAROOTDIR/locale]
–mandir=DIR man documentation [DATAROOTDIR/man]
–docdir=DIR documentation root [DATAROOTDIR/doc/pjproject]
–htmldir=DIR html documentation [DOCDIR]
–dvidir=DIR dvi documentation [DOCDIR]
–pdfdir=DIR pdf documentation [DOCDIR]
–psdir=DIR ps documentation [DOCDIR]

System types:
–build=BUILD configure for building on BUILD [guessed]
–host=HOST cross-compile to build programs to run on HOST [BUILD]
–target=TARGET configure for building compilers for TARGET [HOST]

Optional Features:
–disable-option-checking ignore unrecognized –enable/–with options
–disable-FEATURE do not include FEATURE (same as –enable-FEATURE=no)
–enable-FEATURE[=ARG] include FEATURE [ARG=yes]
Disable floating point where possible
–enable-epoll Use /dev/epoll ioqueue on Linux (experimental)
–enable-shared Build shared libraries
–disable-resample Disable resampling implementations
–disable-sound Exclude sound (i.e. use null sound)
–disable-video Disable video feature
–enable-ext-sound PJMEDIA will not provide any sound device backend
–disable-small-filter Exclude small filter in resampling
–disable-large-filter Exclude large filter in resampling
–disable-speex-aec Exclude Speex Acoustic Echo Canceller/AEC
–disable-g711-codec Exclude G.711 codecs from the build
–disable-l16-codec Exclude Linear/L16 codec family from the build
–disable-gsm-codec Exclude GSM codec in the build
–disable-g722-codec Exclude G.722 codec in the build
–disable-g7221-codec Exclude G.7221 codec in the build
–disable-speex-codec Exclude Speex codecs in the build
–disable-ilbc-codec Exclude iLBC codec in the build
–enable-libsamplerate Link with libsamplerate when available.
–enable-resample-dll Build libresample as shared library
–disable-sdl Disable SDL (default: not disabled)
–disable-ffmpeg Disable ffmpeg (default: not disabled)
–disable-v4l2 Disable Video4Linux2 (default: not disabled)
–disable-openh264 Disable OpenH264 (default: not disabled)
–enable-ipp Enable Intel IPP support. Specify the Intel IPP
package and samples location using IPPROOT and
IPPSAMPLES env var or with –with-ipp and
–with-ipp-samples options
–disable-darwin-ssl Exclude Darwin SSL (default: autodetect)
–disable-ssl Exclude SSL support the build (default: autodetect)

–disable-opencore-amr Exclude OpenCORE AMR support from the build
(default: autodetect)

–disable-silk Exclude SILK support from the build (default:

–disable-opus Exclude OPUS support from the build (default:

–disable-bcg729 Disable bcg729 (default: not disabled)
–disable-libyuv Exclude libyuv in the build
–disable-libwebrtc Exclude libwebrtc in the build

Optional Packages:
–with-PACKAGE[=ARG] use PACKAGE [ARG=yes]
–without-PACKAGE do not use PACKAGE (same as –with-PACKAGE=no)
–with-external-speex Use external Speex development files, not the one in
“third_party” directory. When this option is set,
make sure that Speex is accessible to use (hint: use
CFLAGS and LDFLAGS env var to set the include/lib
–with-external-gsm Use external GSM codec library, not the one in
“third_party” directory. When this option is set,
make sure that the GSM include/lib files are
accessible to use (hint: use CFLAGS and LDFLAGS env
var to set the include/lib paths)
–with-external-srtp Use external SRTP development files, not the one in
“third_party” directory. When this option is set,
make sure that SRTP is accessible to use (hint: use
CFLAGS and LDFLAGS env var to set the include/lib
–with-external-yuv Use external libyuv development files, not the one
in “third_party” directory. When this option is set,
make sure that libyuv is accessible to use (hint:
use CFLAGS and LDFLAGS env var to set the
include/lib paths)
–with-external-webrtc Use external webrtc development files, not the one
in “third_party” directory. When this option is set,
make sure that webrtc is accessible to use (hint:
use CFLAGS and LDFLAGS env var to set the
include/lib paths)
–with-external-pa Use external PortAudio development files. When this
option is set, make sure that PortAudio is
accessible to use (hint: use CFLAGS and LDFLAGS env
var to set the include/lib paths)
–with-sdl=DIR Specify alternate libSDL prefix
–with-ffmpeg=DIR Specify alternate FFMPEG prefix
–with-openh264=DIR Specify alternate OpenH264 prefix
–with-ipp=DIR Specify the Intel IPP location
–with-ipp-samples=DIR Specify the Intel IPP samples location
–with-ipp-arch=ARCH Specify the Intel IPP ARCH suffix, e.g. “64” or
“em64t. Default is blank for IA32”
–with-ssl=DIR Specify alternate SSL library prefix. This option
will try to find OpenSSL first, then if not found,
GnuTLS. To skip OpenSSL finding, use –with-gnutls
option instead.
–with-gnutls=DIR Specify alternate GnuTLS prefix
This option is obsolete and replaced by
–with-opencore-amr=DIR Specify alternate libopencore-amr prefix
Specify alternate libvo-amrwbenc prefix
–with-silk=DIR Specify alternate SILK prefix
–with-opus=DIR Specify alternate OPUS prefix
–with-bcg729=DIR Specify alternate bcg729 prefix

Some influential environment variables:
CC C compiler command
CFLAGS C compiler flags
LDFLAGS linker flags, e.g. -L if you have libraries in a
nonstandard directory
LIBS libraries to pass to the linker, e.g. -l
CPPFLAGS (Objective) C/C++ preprocessor flags, e.g. -I if
you have headers in a nonstandard directory
CXX C++ compiler command
CXXFLAGS C++ compiler flags
CPP C preprocessor

Make and install

make dep
make install


create library instance and initiate with default config and logging

lib = pj.Lib()
lib.init(log_cfg = pj.LogConfig(level=LOG_LEVEL, callback=log_cb))

create UDP transport listening on any available port

transport = lib.create_transport(pj.TransportType.UDP,

create sipuri and local account

my_sip_uri = "sip:" + + ":" + str(
acc = lib.create_account_for_transport(transport, cb=MyAccountCallback())

Function to make call

def make_call(uri):
        print "Making call to", uri
        return acc.make_call(uri, cb=MyCallCallback())
    except pj.Error, e:
        print "Exception: " + str(e)
        return None

Ask user input for destination URI and Call

print "Enter destination URI to call: ",
input = sys.stdin.readline().rstrip("\r\n")
if input == "":
lck = lib.auto_lock()
current_call = make_call(input)
del lck

shutdown the library

transport = None
 acc = None
 lib = None


➜  ~ python sip:altanai@

09:58:09.571        os_core_unix.c !pjlib 2.9 for POSIX initialized
09:58:09.573        sip_endpoint.c  .Creating endpoint instance…
09:58:09.574        pjlib  .select() I/O Queue created (0x7fcfe00590d8)
09:58:09.574        sip_endpoint.c  .Module "mod-msg-print" registered
09:58:09.574        sip_transport.c  .Transport manager created.
09:58:09.575        pjsua_core.c  .PJSUA state changed: NULL --> CREATED
09:58:10.073        pjsua_core.c  .pjsua version 2.9 for Darwin-18.7/x86_64 initialized
Call is  CALLING last code = 0 ()

Debug Help

Isssue1 Undefined symbols for architecture x86_64:
“_pjmedia_codec_opencore_amrnb_deinit”, referenced from:
_amr_encode_decode in mips_test.o
_create_stream_amr in mips_test.o
“_pjmedia_codec_opencore_amrnb_init”, referenced from:
_amr_encode_decode in mips_test.o
_create_stream_amr in mips_test.o
ld: symbol(s) not found for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)
make[2]: *** [../bin/pjmedia-test-x86_64-apple-darwin18.7.0] Error 1
make[1]: *** [pjmedia-test-x86_64-apple-darwin18.7.0] Error 2
make: *** [all] Error 1
*Solution* Mac often throws this linker error. Follow

./configure --enable-shared --disable-static --enable-memalign-hack
make dep
make install
cd pjproject-2.9/pjsip-apps/src/python/
python install

XMPP Client Server Setup and Programming

XMPP is a open XML technology for real-time communication. Applications are instant messaging, presence, media negotiation, whiteboarding, collaboration, lightweight middleware, content syndication, and generalized XML routing according to XMPP standards Foundation (XSF) .

Extensible Messaging and Presence Protocol (XMPP) is a communications protocol for message-oriented middleware based on XML (Extensible Markup Language). – wikipedia

XMPP Server

Some popular servers on XMPP are ejabbred ( written in erlang licensed by GPL2) and openfire ( written in Java licensed by Apache ). This article will show the installation steps for openfire on Ubuntu version 15 64 bit system

1.Install the tar from

Screenshot from 2015-09-25 15:12:02

2. Extract and move the folder to /opt

3. Goto bin and run  openfire server  with ./openfire start

Screenshot from 2015-09-24 12:46:12 (copy)

4. Gotot the web admin url http://localhost:9090/ .  For first time  the setup screen will appear

Screenshot from 2015-09-24 12:46:31

5.  Proceed with installation  .

Screenshot from 2015-09-24 12:46:12

It will show screens to select the mysql driver and database . Create a empty db name called openfiredb and add that to mysql url in setup screen of openfire

It will also request a administrator username and password I choose to give admin admin as the username and password alike .

6. change the interface inside of openfire.xml file in location /opt/openfire/conf


we can also review the mysql connection string

<username encrypted=”true”><<someval>></username>
<password encrypted=”true”> <<someval>></password>
<testSQL>select 1</testSQL>

7. After the installation login to the server admin console with the admin username and password which is admin admin in our case

Screenshot from 2015-09-24 12:54:08

8.  Review the server settings etc from the admin web console

Screenshot from 2015-09-24 13:16:29

9. Incase the server setup did not go as planned we can reinstall the server again by dropping the database , creating a fresh empty database and modifying the following from true to false in openfire.xml file in location /opt/openfire/conf


Test the XMPP Server Installation using Spark client

1.Spark can also be downloaded from the same url as was used to download server . Choose your operating system for download

2.Register a spark client with the server

Screenshot from 2015-09-24 14:41:04

3. after registering the client presence should be indicated in the user summary by online status

Screenshot from 2015-09-25 12:55:13

4.Register another client with the same conf except username and password and perform messaging between them

Screenshot from 2015-09-24 14:45:57

XMPP Java Client

Source Code for a Simple Java Application using Smack4 communicating with XMPP servers

package testxmppsmack;


import org.jivesoftware.smack.ConnectionConfiguration.SecurityMode;
import org.jivesoftware.smack.SmackException;
import org.jivesoftware.smack.XMPPException;
import org.jivesoftware.smack.SmackException.NotConnectedException;
import org.jivesoftware.smack.packet.Message;
import org.jivesoftware.smack.tcp.XMPPTCPConnection;
import org.jivesoftware.smack.tcp.XMPPTCPConnectionConfiguration;

public class JabberSmackAPI {

public static void main(String argsp[]){

XMPPTCPConnectionConfiguration config = XMPPTCPConnectionConfiguration.builder()
.setUsernameAndPassword("admin", "admin")
/* .setSecurityMode(SecurityMode.required) keep this commented */

// Create a connection to the the local XMPP server as defined in config above.
XMPPTCPConnection con = new XMPPTCPConnection(config);

// Connect to the server code is encapsulated in try/catch block for exception handling
try {
System.out.println("Connected "+con.isConnected());
} catch (SmackException | IOException | XMPPException e1) {
// TODO Auto-generated catch block

//Login before performing other tasks like messaging etc
try {
con.login("altanai", "aaa");
System.out.println("Loggedin "+con.isAuthenticated());
} catch (XMPPException | SmackException | IOException e) {
// TODO Auto-generated catch block

// Start a new conversation with another account holder caled altanaibisht ( I created 2 user accounts one with my first name and another with fullname)
Chat chat = ChatManager.getInstanceFor(con).createChat("altanaibisht@localhost");

try {
chat.sendMessage("Did you try out the new code i send you last night ?");
System.out.println("Chat Send ");
} catch (NotConnectedException e) {
// TODO Auto-generated catch block

// Disconnect from the server


Some errors and their resolution while building and running the above code as Java Application are as follows :

1. Cannot instantiate XMPPConnection
Use XMPPTCPConnection instead of XMPPConnection in Smack 4.

2. Caused by: java.lang.ClassNotFoundException: org.xmlpull.v1.XmlPullParserFactory

need to have XPP3 (XML Pull Parser 3) in your classpath. Smack 4 does no longer bundle it (unlike Smack 3).

Download the xpp3 from

ref :

3. Exception in thread “main” java.lang.NoClassDefFoundError: de/measite/minidns/DNSCache

4.  For the jxmpp-util-cache-0.5.0-alpha2.jar

Install it from

5.Exception in thread “main” java.lang.NoClassDefFoundError: org/jxmpp/util/XmppStringUtils

6. Exception in thread “main” java.lang.NoClassDefFoundError: org/apache/http/conn/ssl/StrictHostnameVerifier

7.Exception in thread “main” java.lang.NoClassDefFoundError: org/xbill/DNS/Lookup$ConnectionException: The following addresses failed: ‘machine:5222’ failed because Connection refused


9. org.jivesoftware.smack.SmackException: PKIX path building failed: unable to find valid certification path to requested target


Once the program build and runs succesfully connecting to the XMPP server ( which is running ofcourse ) , open a sapark client and test the application with it.

Screenshot from 2015-09-25 12:44:55


An alternative to XMPP messaging is the SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE) based on Session Initiation Protocol (SIP).

References :

2.Getting started from

3.IETF RFCs on XMPP ( 2004 ) –
RFC 3920
RFC 3921

4. Extensions on XMPP

5. XMPP API explanation by grepcode

XMPP server with JavaScript XMPP clients ( Converse.js)

The setps to install , configure and test a Openfire XMPP server was discussed in my previous blog . It also contained the java client code to interact with the XMPP server like connect , send presence , get and send message etc .

This article will describe the process of making a web based XMPP client that


Converse.js is a free and open-source XMPP chat client written in Javascript . Steps to get the xmpp javascript client working

  1. get the repo access from github

mkdir xmppclient

cd xmppclient

git init

git remote add origin

git fetch

  1. npm install and update

sudo apt-get install npm

npm install -g http-server node-inspector forever nodemon

  1. Make the project
make dev forever in ubuntu


 make -f dev in windows


 npm install bower update 




H 323

International standard for multimedia communication ( real-time voice, video, and data communication) over packet-switched networks
includes the standards H.323, H.225.0, H.245, the H.450-series documents, and the H.460-series.
T.120 for data collaboration and file transfer.
Ability to utilize network services for address resolution, including ENUM, LDAP, and DNS an provide various supplementary services Call transfer , forward , park/pick-up , Hold , Waiting , Message Waiting Indication , Call Completion on Busy / No-Answer , Call Intrusion etc
Support for T.120 data conferencing, fax and modem relay

Elements of an H.323 System

• Terminals ( Telephones , Video phones , IVR , Voicemail Systems, Soft phones like, NetMeeting)
• Multipoint Control Units (MCUs) – manages the call signaling and optionally have Multipoint Processors to handle media mixing, switching or other media processing got multipoint conferences
• Gateways – MGC handles call signaling and other non media-related functions while MG handles the media
• Gatekeeper – admission control and address resolution
• Border Elements

Protocols in H323 framework

H.225.0 defines the call signaling between endpoints and the Gatekeeper
RTP/RTCP (RFC 3550) is used to transmit media such as audio and video over IP networks
H.225.0 Annex G and H.501 define the procedures and protocol for communication within and between Peer Elements
H.245 control establishment and closure of media channels within the context of a call and to perform conference control
H.450.x is a series of supplementary service protocols
H.460.x is a series of version-independent extensions to the base H.323 protocol
T.120 specifies how to do data conferencing
T.38 defines how to relay fax signals
V.150.1 defines how to relay modem signals
H.235 defines security within H.323 systems
X.680 defines the ASN.1 syntax used by the Recommendations
X.691 defines the Packed Encoding Rules (PER) used to
encode messages for transmission on the network

Ref :