Month: May 2014

Internet Telephony Convergence- JAINSLEE Platform

Convergence : Telephone networks and computer networks converging into single digital network using Internet standards.

Components in a Network

  • Client computer
  • Server computer
  • Network interfaces (NICs)
  • Connection medium
  • Network operating system
  • Hub or switch
  • Routers- Device used to route packets of data through different networks, ensuring that data sent gets to the correct address

1

Figure :simple computer network, consisting of computers, a network operating system residing on a dedicated server computer, cable (wiring) connecting the devices, network interface cards (NICs), switches, and a router.

2

Figure of a Enterprise Network : local area networks (LANs) linked to enterprise level / corporate network . It consists of :

  • Powerful servers
  • Web site
  • Corporate intranet, extranet
  • Backend systems
  • Mobile wireless LANs (Wi-Fi networks)
  • Videoconferencing system
  • Telephone network
  • Wireless cell phones etc

The convergence of Internet and Telephony opens up new revenue streams for the Communication Service Providers by delivering new innovation based convergent applications.

Before discussing digitization of Communication and telecom we need to first understand packet switching .

What is Packet switching ?
It is a method of slicing digital messages into packets, sending them along different communication paths as they become available. Then reassembling these received packets at destination. It is a more efficient use of network’s communications capacity.
Previous circuit-switched networks required assembly of complete point-to-point circuit.

What triggered this Technology development?

The Internet, IPTV and Social Media networking is evolving dynamically in the end user space of Communication Service Provider. This opens door for delivering new innovative services to end user through these converged applications.

A SP( Service provider ) has to work with multiple Communication Providers globally and based on the experience with the customers, has to conceptualize and implemented new innovative use cases on open platform to reduce the cost and  migrate from legacy to Next Generation Networks.

What does convergence mean to

  • Equipment Vendors / EV
    • Femto / FMC
    • Challenges in System Integration
    • Box to Solution Sales
    • Services software based
  • Enterprises
    • Low Capex – Hosted Models
    • Enterprise Mobility
    • IP Enabled Services
    • UC to UC
    • Web Integration with Mobility
  • Telco
    • Enterprise communication will be a big focus Area
    • Push to EVs for CAPEX
    • Wish to leverage Legacy as well
    • Challenges in Vertical Solutions
    • Will face challenges by OTT players

 

What will it do, how and in which situation ?

The underlying technology of Internet Telephony Convergent Platform is JAIN SLEE Framework which is open standard for developing core network based applications. It enables development of network agnostic applications , implemented through resource adapters for deploying same applications over different networks like SIP/IN etc.

JAINSLEE framework provides capability to form new complex services through reusable service building block in much easier way then traditional methods. This reduces cost for launching new services and bundled different services into the new convergent service in network agnostic way. 

It also bring benefits in term of reducing the dependency on Vendor proprietary platform and eventually bringing down cost involved and Time to market in launching new service.

The OpenCloud Service Layer OpenCloud

picture courtesy : Opencloud.com

What problem does this technology seek to solve?

Today communication service provider are facing vendor locking situation where most of services deployed are platform dependent which requires huge cost of investment for launching new services. Traditional service development platforms are major roadblock for operators to launch new collaborative services which involves both voice and data channels as they are not based on open standards and are tied to the vendor specific technologies. Also in a fast changing technology the operators need to switch their focus on new innovative services through which operator can monetize services and provide the value added experience to their end customers. To enable it we proposed and implemented framework which not only act as the new Internet Telephony convergent platform but also in sync with their future network transformation strategy as it is based on open standards. Through this platform same applications can be targeted to different segment of users with minimal cost impact. Some of the application which we have developed are detailed below.

a) Parental Control is an application through which parents can have control over their children’s Internet video on demand request. Once a child requests for any video, preview of the same(short clip of video) at the same instance is send to parents’ smart phones. Parents can see preview and can decide there and then weather it is adequate for his/her kids or not, and can either allow or deny through his mobile.

b) IPTV/VOD session mobility is a service which allow user to transfer their ongoing voice call/video-on-demand session from their smartphone to desktop/computing device/smart-device and vice-versa seamlessly.

c) Converged application like unified communication platform for trader community take advantage of both voice and data services and help trader community in terms of analytics and decision making process.

What is the specific breakthrough of this technology?

Internet and Telephony are two major drivers in Telecom domain. Hence the concept of convergence of Internet and Telephony is of great interest for the Telcos. Internet telephony, also known as voice-over-IP or IP telephony is the real-time delivery of voice between two or more parties, across networks using the Internet protocols, and the exchange of information required to control this delivery. New innovative use case scenarios  have been conceptualized and implemented considering new user behavior changes. These bring in value addition to CSPs in order to bring more revenue streams. Solutions like Secure VOIP bring another dimension of innovation as it provides a secured voice communication over the internet using open source software like Asterisk. This solution helps business reduce their operational communication costs using encrypted standard security algorithms.

Asterisk- Applications (1)

pic courtesy : asterisk.com

How does this technology compare with other technologies? 

Internet telephony convergent platform has the unique value proposition based on new innovative use case scenarios using multiple underlying technologies. These scenarios are implemented using Open Standards. Though many other vendors’ platform also provides some of the facilities of platform in part and pieces but none of them give complete end to end solutions suits to operators as our Internet Telephony convergent platform provides.

How does it help in achieving the goals?

We consider it as solution which can act as foundation block to build a long term partnership with operators especially in area of services landscape. This solution enables operator to monetize different voice and data convergent services and in sync with the operator’s next generation transformation initiative. The services acts as catalyst to increase the data usage of end-users. Strong business case can be built with these services by operators as they meet the future demands of tech savvy end users. These services not only fill the void between communication service provider and social media/internet/video-on-internet but also take advantage of reach of social media/internet and eventually enable operator to add new revenue stream. These services can also help operator to increase their brand visibility with added advantage of social media and internet application bundled with their core services. Operator can charge it on per application basis or can be just carrier and charge for data usage. Convergent services which involves both the voice and data, enable operator to charge on voice services , data services and application usage. With our rich experience in convergent platform domain we believe we can convert significant opportunities in this space.

Explain your journey of Technology development ?

After seeding of concept of Internet Telephony convergent platform SP should explore partner product Software centric platforms like Open cloud, Oracle, Mobicient etc which offers the capability to deliver convergent applications at a low cost and using the open standards. Standards like JAIN SLEE provide capability for developing and delivering such applications across different type of underlying network. 

Mobicents Platform

pic courtesy : Mobicents.com

One can develop the complete solution using such open, standard platforms as a base . The complete solution takes care of the real-network issues and solutions for the same. There were many hurdles and roadblock at first. Adaptation to open standards like JAIN SLEE requires fast ramp up as it is quite complex technology. In a small stipulated time a core team should have developed competency through Partner Training inputs and Brain Storming sessions. To test framework at lab, there would be dependency on many open source software and strategic partner products. There would be many incompatibility issues. Its important that such issues be  sorted out by exhaustive explorations of products and by bug fixes .

Benefits expected if this Technology is implemented / commercialized 

a) Communication service providers are able to realize appreciable cost saving through Internet Telephony convergent platform Operators deployed in their network. This is so legacy platform were costly and difficult to manage. This platform brings innovative and cost effective way of launching new collaborative services which brings new revenue stream.

b) Improved Time to market

c) Extensible architecture for the service helps in extending the service for multiple markets.

Social Benefits

Unified communications, where voice, video, email, text and other messaging technologies are combined to provide greater flexibility for users by enabling new ways to transfer information and manage connectivity. Integration of collaborative services with the social media platform like Facebook , Linkedin , Twitter etc, increases the connectivity and value experience of end users. Through social media based convergent applications operator can further increase their reach to end users by utilizing underlying the Internet Telephony convergent platform.

My Insights 

Based on my personal experience while implementing this technology/platform, I think this solution act as catalyst for enabling the transition from network eccentricity to customer eccentricity. This movement is further supplemented through the reduced dependence on legacy vendors and increased adoption of open standard based platforms. Through the converged application layer for Telcos I envisage a platform which is agnostic to underlying network layer. Unified platform allows carriers, mobile operators, and cable operators to rapidly create, manage, and deliver converged video, voice, and data service bundles across multiple networks and devices. It enhance end user experience and enable Telcos to add new revenue stream by offering value added services to their customer. 

Kamailio Call routing and Control

Kamailio SIP server evolved from SER and OpenSER. Written in ANSI C , primarily it is an open source proxy SIP server. RFC 3261 compliant and has support for various Operating system to install and run on as alpine , centos , deb , fedora , freebsd , netbsd , obs , openbsd , opensuse , oracle , rhel , solaris so on .

With modular design it already has 150 + modules and can have third party addons like Databases , RTP engines etc. Anyone can contribute to extensions and modules read here. Also contains cmd line tool kamcmd , kamcli and Web management interface SIREMIS .

It has provisions for complex routing logic development through scripts and programming languages interpreter support.

Over the years kamailio as proven a key component of a “carrier-grade” SIP service delivery platform. Either as SBC interfacing internal softswitch with public internet and handling complex operation as NAT, auth , flood control, topology hiding etc or even as the core SIP Server handling RTP relay as well.

Kamailio’s call routing log can be divided into:

  • Core functions
  • Core modules needed for almost any useful configuration like transaction module , Record route module etc ( read more https://telecom.altanai.com/2014/11/18/kamailio-modules/)
  • Ancillary modules to provide specific functionality (e.g. JANSSON , PIKE)
  • Pseudo-variables , which hold read/writeable attributes and
  • Transformations and custom logic definition.

Kamailio config

kamailio confguration file should be composed of
Core configuration directives as

!ifdef WITH_MULTIDOMAIN
!define MULTIDOMAIN 1
!else
!define MULTIDOMAIN 0
!endif

Global Parameters like

memdbg=5
memlog=5
log_facility=LOG_LOCAL0
fork=yes
children=4

Loading Modules

loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"

Module parameters
For example considering for tm auto-discard branches from previous serial forking leg as failure_reply_mode ,30 sec as default retransmission timeout with 120 sec as invite retransmission timeout after 1xx

modparam("tm", "failure_reply_mode", 3)
modparam("tm", "fr_timer", 30000)
modparam("tm", "fr_inv_timer", 120000)

Subroutines (in essence, SIP event callbacks):
Request routes (request_route)
Reply routes (onreply_route)
Failure routes (failure_route)
Branch routes

For example consider main sip routing block

request_route {
    route(REQINIT);
    route(NATDETECT);
    if (is_method("CANCEL"))
    {
        if (t_check_trans()) {
            route(RELAY);
        }
        exit;
    }
    route(WITHINDLG);
    t_check_trans();
    route(AUTH);
    if (is_method("INVITE|SUBSCRIBE"))
        record_route();
    route(SIPOUT);
    route(PRESENCE);
    route(REGISTRAR);
    ...
}

Custom event routes (callbacks/event handlers exposed by modules).
Code for programming languages and runtimes:
String transformations
Variables
Ephemeral/scratch-pad variables ($var(…))
Transaction-persistent variables ($avp(…)/$xavp(…)) , extended AVP like AVP ar attached to transactions and not messages .
Dialog-persistent variables ($dlg_var(…))

$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
    case -1:
    case -3:
        send_reply("404", "Not Found");
        exit;
    case -2:
        send_reply("405", "Method Not Allowed");
        exit;
}

This article describes call routing config for Kamailio under following roles

  • SIP Proxy
  • Registrar
  • Accountant
  • Session border Controller

Kamailio as Proxy Server

Simple Kamailio configuration with basic features like alias , accounting , record routing , handling SIP requests like INVITE and its replies . Also failure and NAT handling . More samples of Kamailio config and call routing are at https://github.com/altanai/kamailioexamples

#!KAMAILIO

#Defined Values
!substdef "!MY_IP_ADDR!!g"
!substdef "!MY_EXTERNAL_IP!!g"
!substdef "!MY_UDP_PORT!!g"
!substdef "!MY_TCP_PORT!!g"
!substdef "!MY_UDP_ADDR!udp:MY_IP_ADDR:MY_UDP_PORT!g"
!substdef "!MY_TCP_ADDR!tcp:MY_IP_ADDR:MY_TCP_PORT!g"
!define MULTIDOMAIN 0

; - flags
; FLT_ - per transaction (message) flags
; FLB_ - per branch flags
!define FLT_ACC 1
!define FLT_ACCMISSED 2
!define FLT_ACCFAILED 3
!define FLT_NATS 5
!define FLB_NATB 6
!define FLB_NATSIPPING 7

# Global Parameters
; LOG Levels:3 = DBG, 2 = INFO, 1 = NOTICE, 0 = WARN, -1 = ERR
debug = 2
log_stderror = no
memdbg = 5
memlog = 5
log_facility = LOG_LOCAL0
log_prefix = "{$mt $hdr(CSeq) $ci} "

/* number of SIP routing processes */
children = 2
/* uncomment the next line to disable TCP (default on) */
disable_tcp = yes
/* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */
auto_aliases = no

/* add local domain aliases */
alias = "sip.mydomain.com"

/* listen addresses */
listen = udp:127.0.0.1:5060
listen = MY_UDP_ADDR advertise MY_EXTERNAL_IP:MY_UDP_PORT
listen = MY_TCP_ADDR advertise MY_EXTERNAL_IP:MY_TCP_PORT

# Modules Section
loadmodule "jsonrpcs.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "acc.so"
loadmodule "counters.so"

----------------- setting module-specific parameters --------------

----- jsonrpcs params -----
modparam("jsonrpcs", "pretty_format", 1)
/* set the path to RPC fifo control file */
modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo")
/* set the path to RPC unix socket control file */
modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock")

; ----- ctl params -----
/* set the path to RPC unix socket control file */
modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")

; ----- tm params -----
auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
default retransmission timeout:30sec
modparam("tm", "fr_timer", 30000)
default invite retransmission timeout after 1xx:120sec
modparam("tm", "fr_inv_timer", 120000)

; ----- rr params -----
# set next param to 1 to add value to;lr param (helps with some UAs)
modparam("rr", "enable_full_lr", 0)
; do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)

----- acc params -----
; /* what special events should be accounted ? / modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) / by default ww do
; not adjust the direct of the sequential requests.
; if you enable this parameter, be sure the enable "append_fromtag"
; in "rr" module / 
modparam("acc", "detect_direction", 0) / account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)

# Routing Logic

/* Main SIP request routing logic*/
request_route {
;     per request initial checks
    route(REQINIT);
    ; CANCEL processing
    if (is_method("CANCEL")) {
        if (t_check_trans()) {
            route(RELAY);
        }
        exit;
    }
    
    ; handle retransmissions
    if (!is_method("ACK")) {
        if (t_precheck_trans()) {
            t_check_trans();
            exit;
        }
        t_check_trans();
    }

    ; handle requests within SIP dialogs
    route(WITHINDLG);
    ;     only initial requests (no To tag)
    ; record routing for dialog forming requests ( in case they are routed)
    ; - remove preloaded route headers
    remove_hf("Route");
    if (is_method("INVITE|SUBSCRIBE")) {
        record_route();
    }
    
    ;     account only INVITEs
    if (is_method("INVITE")) {
        setflag(FLT_ACC); # do accounting
    }
    
    if ($rU==$null) {
        # request with no Username in RURI
        sl_send_reply("484", "Address Incomplete");
        exit;
    }

    # update $du to set the destination address for proxying
    $du = "sip:" + $rd + ":9";
    route(RELAY);
    exit;
}

# Wrapper for relaying requests
route[RELAY] {
    if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {     
        if (!t_is_set("branch_route")) 
            t_on_branch("MANAGE_BRANCH");
    }

    if (is_method("INVITE|SUBSCRIBE|UPDATE")) {     
        if (!t_is_set("onreply_route")) 
            t_on_reply("MANAGE_REPLY");
    }

    if (is_method("INVITE")) {
        if (!t_is_set("failure_route")) 
            t_on_failure("MANAGE_FAILURE");
    }

    if (!t_relay()) {     
        sl_reply_error(); 
    } 
    exit;
}

#P er SIP request initial checks
route[REQINIT] {
    if ($ua = ~ "friendly-scanner|sipcli|VaxSIPUserAgent") {
        # sl_send_reply("200", "OK");
        exit;
    }
    
    if (!mf_process_maxfwd_header("10")) {
        sl_send_reply("483", "Too Many Hops");
        exit;
    }

    if (is_method("OPTIONS") && uri==myself && $rU==$null) {     
        sl_send_reply("200", "Keepalive");
        exit;
    }

    if (!sanity_check("1511", "7")) {
        xlog("Malformed SIP message from $si:$sp\n");
        exit;
    }
}

# Handle requests within SIP dialogs

route[WITHINDLG] {
    if (!has_totag()) 
        return ;
    
    if (loose_route()) {
        if (is_method("BYE")) {
            setflag(FLT_ACC); # do accounting ...         
            setflag(FLT_ACCFAILED); # ... even if the transaction fails
        } else{
            if (is_method("NOTIFY")) {
                # Add Record-Route for in -dialog NOTIFY as per RFC 6665.         
                record_route();
            }
            route(RELAY);
            exit;
        }
    }   

    if (is_method("ACK")) {
        if (t_check_trans()) {         
            # no loose-route, but stateful ACK;         
            must be an ACK after a 487 or e.g. 404 from upstream server         
            route(RELAY);
            exit;
        } else {
            # ACK without matching transaction, ignore and discard         
            exit;
        }
    } 
    
    sl_send_reply("404", "Not here"); exit;

    #Manage outgoing branches
    branch_route[MANAGE_BRANCH] {
    xdbg("new branch [$T_branch_idx] to $ru\n");
}

--# Manage incoming replies
onreply_route[MANAGE_REPLY] {
    xdbg("incoming reply\n");
}

--# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
    if (t_is_canceled()) exit;
}






Controlling Call Routing from command Line 





To find the loaded flags while starting kamailio 




kamailio -v
version: kamailio 5.1.8 (x86_64/linux) d8e930
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: d8e930 
compiled on 09:47:09 May  3 2019 with gcc 4.8.4





kamdbctl





creates the database support for many kamailio modules such as auth , location , dispatcher , permission etc 





make sure you load a DB engine , during kamailio installation and configuration . It can be either done though make command or though modules.lst file




 make include_modules="db_mysql" cfg
 make all
 make install





since json replaced all fifo command, ensure you do not get  "json.h: No such file or directory” in server by install json either via libjson-c-dev or  libjson-cpp-dev





apt-get install libjson-c-dev 





For uuid/uuid.h: No such file or directory install 





apt-get install uuid-dev





For libpq-fe.h: No such file or directory install





apt-get install libpq-dev





kamdbctl command list




kamdbctl create <db name or db_path, optional> ...(creates a new database)
kamdbctl drop <db name or db_path, optional> .....(!entirely deletes tables!)
kamdbctl reinit <db name or db_path, optional> ...(!entirely deletes and than re-creates tables!)
kamdbctl backup <file> ...........................(dumps current database to file)
kamdbctl restore <file> ..........................(restores tables from a file)
kamdbctl copy <new_db> ...........................(creates a new db from an existing one)
kamdbctl presence ................................(adds the presence related tables)
kamdbctl extra ...................................(adds the extra tables)
kamdbctl dbuid ...................................(adds the uid tables)
kamdbctl dbonly ..................................(creates empty database)
kamdbctl grant ...................................(grant privileges to database)
kamdbctl revoke ..................................(revoke privileges to database)
kamdbctl add-tables <gid> ........................(creates only tables groupped in gid)

if you want to manipulate database as other database user than
root, want to change database name from default value "kamailio",
or want to use other values for users and password, edit the
"config vars" section of the command kamdbctl.

kamdbctl pframework create .......................(creates a sample provisioning framework file)





For psql: received invalid response to SSL negotiation: [
 ERROR: Creating database failed! errors  . Remember for mysql the defaul port is 3306, but for psql it is 5432 . Hence make the change in /etc/kamailio/kamctlrc





database port
DBPORT=3306
DBPORT=5432  





Kamctl





If kamctl isnt accessible from the machine installed with kamailio , just goto kamctl folder and compile it yourself . For example for me , I took the git pull of kamailio source code v 5.1.0 and went to util folder 





cd  /kamailio_source_code/utils/kamctl 
make && make install 





some commands





‘start|stop|restart|trap’

‘acl’ – manage access control lists (acl) 

‘lcr’ – manage least cost routes (lcr) 

‘cr’ – manage carrierroute tables show|reload|dump

‘rpid’ – manage Remote-Party-ID (RPID) 

‘add|passwd|rm’ – manage subscribers 

‘add|dump|reload|rm|show’ – manage trusted

‘add|dump|reload|rm|show’ – manage address

‘add|dump|reload|rm|show’ – manage address

‘dispatcher’ – manage dispatcher




dispatcher add 1 sip:1.2.3.1:5050 1 5 'prefix=proxycall' 'gatewaye33'





‘dialog’ – manage dialog records

‘srv’ – server management commands

‘cisco_restart’ – restart CISCO phone (NOTIFY)

‘online’ – dump online users from memory

‘monitor’ – show internal status




[cycle #: 3; if constant make sure server lives]
Kamailio Runtime Details: 
kamailio 5.1.8 (x86_64/linux) d8e930
    now:  Fri May 24 13:39:19 2019
    up_since: Fri May 24 13:31:37 2019
    uptime: 462

Transaction Statistics: 
    tmx:UAS_transactions = 0        tmx:UAC_transactions = 0        tmx:active_transactions = 0     tmx:inuse_transactions = 0

Stateless Server Statistics: 
    sl:sent_replies = 0            sl:sent_err_replies = 0

UsrLoc Statistics: 
    usrloc:location-contacts = 0        usrloc:location-expires = 0         usrloc:location-users = 0           usrloc:registered_users = 0

Core Statistics: 
    core:rcv_requests = 0        core:fwd_requests = 0       core:rcv_replies = 0        core:fwd_replies = 0

Shared Memory Statistics: 
    shmem:fragments = 1                shmem:max_used_size = 2807640       shmem:total_size = 67108864
    shmem:free_size = 64301224            shmem:real_used_size = 2807640      shmem:used_size = 2566040





‘ping’ – ping a SIP URI (OPTIONS)

‘ul|alias’ – manage user location or aliases

‘ps’ – print details about running processes

‘ps’ – print details about running processes

‘stats’ – print internal statistics




{
  "jsonrpc":  "2.0",
  "result": [
    "core:bad_URIs_rcvd = 0",
    "core:bad_msg_hdr = 0",
    "core:drop_replies = 0",
    "core:drop_requests = 0",
    "core:err_replies = 0",
    "core:err_requests = 0",
    "core:fwd_replies = 0",
    "core:fwd_requests = 0",
    "core:rcv_replies = 0",
    "core:rcv_replies_18x = 0",
    "core:rcv_replies_1xx = 0",
    "core:rcv_replies_2xx = 0",
    "core:rcv_replies_3xx = 0",
    "core:rcv_replies_401 = 0",
    "core:rcv_replies_404 = 0",
    "core:rcv_replies_407 = 0",
    "core:rcv_replies_480 = 0",
    "core:rcv_replies_486 = 0",
    "core:rcv_replies_4xx = 0",
    "core:rcv_replies_5xx = 0",
    "core:rcv_replies_6xx = 0",
    "core:rcv_requests = 0",
    "core:rcv_requests_ack = 0",
    ...
}





‘rpc’ – send raw RPC commands





Kamcmd 





unix tool for interfacing with Kamailio using exported RPCs.
It uses binrpc (a proprietary protocol, designed for minimal packet size and
fast parsing) over a variety of transports (unix stream sockets, unix datagram
 sockets, udp or tcp).





 cfg.add_group_inst
 cfg.commit
 cfg.del
 cfg.del_delayed
 cfg.del_group_inst
 cfg.diff
 cfg.get
 cfg.help
 cfg.list
 cfg.reset
 cfg.rollback
 cfg.set
 cfg.set_delayed
 cfg.set_delayed_int
 cfg.set_delayed_string
 cfg.set_now_int
 cfg.set_now_string
 cfg.seti
 cfg.sets
 cnt.get
 cnt.get_raw
 cnt.get_vars
 cnt.grp_get_all
 cnt.grps_list
 cnt.help
 cnt.list_groups
 cnt.list_vars
 cnt.reset
 cnt.var_list
 core.aliases_list
 core.arg
 core.echo
 core.flags
 core.info
 core.kill
 core.modules
 core.ppdefines
 core.printi
 core.prints
 core.ps
 core.psx
 core.pwd
 core.shmmem
 core.sockets_list
 core.tcp_info
 core.tcp_list
 core.tcp_options
 core.udp4_raw_info
 core.uptime
 core.version
 corex.debug
 corex.list_aliases
 corex.list_sockets
 corex.pkg_summary
 corex.shm_status
 corex.shm_summary
 ctl.connections
 ctl.listen
 ctl.who
 dns.add_a
 dns.add_aaaa
 dns.add_srv
 dns.debug
 dns.debug_all
 dns.delete_a
 dns.delete_aaaa
 dns.delete_all
 dns.delete_all_force
 dns.delete_cname
 dns.delete_ebl
 dns.delete_naptr
 dns.delete_ptr
 dns.delete_srv
 dns.delete_txt
 dns.lookup
 dns.mem_info
 dns.view
 dst_blacklist.add
 dst_blacklist.debug
 dst_blacklist.delete_all
 dst_blacklist.mem_info
 dst_blacklist.view
 jsonrpc.echo
 mod.stats
 pkg.stats
 pv.shvGet
 pv.shvSet
 sl.stats
 stats.clear_statistics
 stats.fetch
 stats.get_statistics
 stats.reset_statistics
 system.listMethods
 system.methodHelp
 system.methodSignature
 tm.cancel
 tm.clean
 tm.hash_stats
 tm.list
 tm.reply
 tm.reply_callid
 tm.stats
 tm.t_uac_start
 tm.t_uac_wait
 ul.add
 ul.db_contacts
 ul.db_expired_contacts
 ul.db_users
 ul.dump
 ul.flush
 ul.lookup
 ul.rm
 ul.rm_contact
 alias: ps
 alias: psx
 alias: list
 alias: ls
 alias: ver
 alias: version
 alias: who
 alias: listen
 alias: dns_mem_info
 alias: dns_debug
 alias: dns_debug_all
 alias: dst_blacklist_mem_info
 alias: dst_blacklist_debug
 builtin: ?
 builtin: help
 builtin: version
 builtin: quit
 builtin: exit
 builtin: warranty
 builtin: license





some examples of the kamcmd commands





stats 





> kamctl kamcmd 
tm.stats 
{     
current: 0     
waiting: 0     
total: 0     
total_local: 0     
rpl_received: 0     
rpl_generated: 0     
rpl_sent: 0     
6xx: 0     
5xx: 0     
4xx: 0     
3xx: 0     
2xx: 0     
created: 0     
freed: 0     
delayed_free: 0 
}





Get info about TLS




kamcmd> tls.info
{
    max_connections: 2048
    opened_connections: 1
    clear_text_write_queued_bytes: 0
}





Get info about open sockets 




kamcmd> core.sockets_list
{
    socket: {
        proto: udp
        address: 1.2.3.4
        port: 5060
        mcast: no
        mhomed: no
    }
    socket: {
        proto: tcp
        address:  1.2.3.4
        port: 5060
        mcast: no
        mhomed: no
    }
    socket: {
        proto: tcp
        address:  1.2.3.4
        port: 80
        mcast: no
        mhomed: no
    }
    socket: {
        proto: tls
        address:  1.2.3.4
        port: 5061
        mcast: no
        mhomed: no
    }
    socket: {
        proto: tls
        address:  1.2.3.4
        port: 443
        mcast: no
        mhomed: no
    }
}





get core info




kamcmd > core.info
{
    version: kamailio 5.2.3
    id: 4a4588 
    compiler: gcc 5.4.0
    compiled: 15:54:50 Jun 27 2019
    flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
}








			


	



			
				


	

		
Business Challenges for a telecom service provider

			


	

		
With the fast pace of telecom evolution both towards the access network front ( ie GSM , UMTS , 3G , 4G , LTE , VOLTE ) to core network side ( ie application servers , registrar , proxies , gateway , media server etc ) a CSP ( content service provider ) is trying hard to keep up with the user expectation . The user expects a plethora of services , reduced cost and high speed bandwidth . If this was not enough a CSP also has competition  OTT (   Over The Top ) Players who provide communication and messaging for FREE .


You can read on how OTT’s players are disruption the revenue streams of traditional telecom operators and how can Telco’s develop  their own OTT app , integrated with their backend system to answer to that challenge  here – OTT ( Over the Top ) Communication applications


The following points outline the major business challenges faced by telecom operators today .


Technology Evolution Challenges


    

  •  The increased data speeds and further more increasing hunger for the data overwhelms the existing network infrastructures.
    • 

  • Ensure uniform service experience across the network technologies to check the customer churn.
    • 

  • Access / Radio Technology independent delivery of services.
    • 

  • Enhance Reuse for exiting investments.
    • 



Multiple Service Platform Challenges


    

  • Typical network constitutes of Multiple Service Platforms increasing network complexity and integration challenges many fold.
    • 

  • Heterogeneous multiple SDP Solutions typically deployed to cater to Multiple Types of Networks/ Standards/Variants
    • 

  • Service Islands makes introduction of seamless services a challenging task for the CSP
    • 



Transport Upgrade and Convergence of Wireless Wireline


    

  • Retain investments in copper wire systems while migrating towards next generation Fiber Optic systems.
    • 

  • Severe competition among wire-line and wireless operators to provide latest services to retain subscriber base.
    • 

  • Fixed Mobile Convergence leading to a diminishing gap among the revenue shares of various operators in the space, and leading to losses for wire-line only players.
    •