This article describes various Certificates and compliances, Bill and Acts on data privacy, Security and prevention of Robocalls as adopted by countries around the world pertaining to Interconnected VoIP providers, telecommunications services, wireless telephone companies etc
Compliance certificates by Industry types
HIPAA (Health Insurance Portability and Accountability Act)
Deals with privacy and security of personal medical records and electronic health care transaction
Applicability : If voip company handles medical information
Includes :
Not allowed Voice mail transcription
Should have End-to-End Encryption
Restrict using unsecured WiFi networks to prevent Snooping
User security , strong password rules and mandatory monthly change
Secure Firmware on VoIP phones
Maintaining Call and Access Logs
SOX( Sarbanes Oxley Act of 2002)
Also known as SOX, SarbOX or Public Company Accounting Reform and Investor Protection Act
Applicability : if managing the communications operations of a regulated, publicly traded company
Includes :
Retain records which include financial and other sensitive data
ways employees are provided or denied access to records or data based on their roles and responsibilities
do information audit by a trusted third party.
Retention and deletion of files such as audio files like voicemails, text messages, video clips, declared paper records, storage, and logs of communications activities
Physical and digital security controls around cloud-based VoIP applications and the networks
Privacy Related Compliance certificates
COPPA (Children’s Online Privacy Protection Act ) of 1998
prohibits deceptive marketing to children under the age of 13, or collecting personal information without disclosure to their parents.
any information is to be passed on to a third party, must be easy for the child’s guardian to review and/or protect
2011 amendment requires that the data collected was erased after a period of time,
2014 FTC issued guidelines that apps and app stores require “verifiable parental consent.”
CPNI (Customer Proprietary Network Information) in united states is the information that communication providers acquire about their subscribers. This Individually identifiable information that is created by a customer’s relationship with a provider, such as data about the frequency, duration, and timing of calls, the information on a customer’s bill, and call identifying information. This processing information is governed strictly by FCC and certification should be renewed on an annual basis
Provider can pass along that information to marketers to sell other services, as long as the customer is notified
In 2007, the FCC explicitly extended the application of the Commission’s CPNI rules of the Telecommunications Act of 1996 to providers of interconnected VoIP service.
CALEA
Communications Assistance for Law Enforcement Act (CALEA) conduct electronic surveillance by imposing specific obligations on “telecommunications carriers” for assisting law enforcement, including delivering call interception and call identification functionality to the government with a minimum of interference to customer service and privacy.
Establishes requirements of organizations that process data, defines the rights of individuals to manage their data, and outlines penalties for those who violate these rights.
No personal data may be processed unless this processing is done under one of six lawful bases specified by the regulation (consent, contract, public task, vital interest, legitimate interest or legal requirement). When the processing is based on consent the data subject has the right to revoke it at any time.
Controllers must notify Supervising Authorities (SA)s of a personal data breach within 72 hours of learning of the breach.
California Consumer Privacy Act (CCPA) 2019
consumer rights relating to the access to, deletion of, and sharing of personal information that is collected by businesses.
Allows consumers to know whether their personal data is sold or disclosed , to whom .
Allows opt-out right for sales of personal information
Right to deletion – to request a business to delete any personal information about a consumer collected from that consumer
Personal Data Protection Bill (PDP) – India 2018
This bill introduces various private and sensitive protection frameworks like restriction on retention of personal data, Right to correction and erasure (such as right to be forgotten) , Prohibition and transparency of processing of personal data. It also classifies data fiduciaries including certain social media intermediaries.
The Bill amends the Information Technology Act, 2000 to delete the provisions related to compensation payable by companies for failure to protect personal data.
Other data privacy acts similar to GDPR
South Korea’s Personal Information Protection Act 2011
Brazil’s Lei Geral de Proteçao de Dados (LGPD) 2020
Privacy Amendment (Notifiable Data Breaches) to Australia’s Privacy Act 2018
Japan’s Act on Protection of Personal Information 2017
Thailand Personal Data Protection Act (PDPA) 2020
Features offered by VOIP companies for Data privacy
Access Control & Logging
Auto Data Redaction / Account Deletion policy
SIEM (Security information and event management) alerts
Information security , Encrypted Storage For Recordings & Transcripts
Disclosing all third party services that are involved in data processing too
Role Based Access Control and 2 Factor Authentication
Data Security Audits and appointing data protection officer to oversee GDPR compliance
Against Robocalls and SPIT ( SPAM over Internet Telephony)
2009 Truth in Caller ID Act
Telephone Consumer Protection Act of 1991
Implementation of Do not call registry against use of robocalls, automatic dialers, and other methods of communication
Do-Not-Call Implementation Act of 2003
if a business has an established relationship with a customer, it can continue to call them for up to 18 months. If a consumer calls the company, say, to ask for information about the product or service, the company has three months to get back to him.
if the customer asks to not receive calls, the company must stop calling, or be subject to fines.
Exemptions – Calls from a not-for-profit B organisation , informational messages as flight cancellations , Calls from sales and debt collectors etc
Personal Data Privacy and Security Act 2009
Implemented to curb identity theft and computer hacking. Sensitive personal identifiable information includes : victim’s name, social security number, home address, fingerprint/biometrics data, date of birth, and bank account numbers.
Any company that is breached must notify the affected individuals by mail, telephone, or email, and the message must include information on the company and how to get in touch with credit reporting agencies
If the breach involves government or national security , company must also contact the Secret Service within fourteen days
TRACED Act (Telephone Robocall Abuse Criminal Enforcement and Deterrence) 2019
Canadian Radio-television and Telecommunications Commission (CRTC) 2018 -32
Unlike traditional telephone connections, which are tied to a physical location, VOIP’s packet switched technology allows a particular number to be anywhere making it more difficult for it to reach localised services like emergency numbers of Public Safety Answering Points (PSAPs) . Thus FCC regulations as well as the New and Emerging Technologies 911 Improvement Act of 2008 (NET 911 Act), interconnected VoIP providers are required to provide 911 and E911 service.
To understand the need for implementing an identification verification technique in Internet protocol based network to network communication system , we need to evaluate the existing problem plaguing the VoIP setup .
What is Call ID spoofing ?
Vulnerability of existing interconnection phone system which is used by robo-callers to mask their identity or to make it appear the call is from a legitimate source, usually originates from voice-over-IP (VOIP) systems.
In this context understand the Caller Line identification CLI/ NCLI techniques used by VoIP and OTT( over the top) providers today.
CLI (Caller Line Identification)
If call goes out on a CLI route ( White Route ) the received party will likely see your callerID information
Lawful – Termination is legal on the remote end ie abiding country’s telco infrastructure and stable
Expensive – usually with direct or via leased line (TDM) interconnections with the tier-1 carriers.
Non-CLI (Non-Caller Line Identification)
The Caller ID is not visible at the call If call goes out on a Non-CLI route (Grey Route) goes out on a non-CLI routes they will see either a blocked call or some generic number.
Unlawful – questionable legality or maybe violating some providers AUP(Acceptable Use Policy ) on the remote end.
Cheaper – low quality , usually via VoIP-GSM gateways
Example include robocalls , tele-marketting / spam etc which are unwilling to share their Caller Id for call receiver, to not be blocked or cancelled.
To overcome the problem of non-verifiable spam , robocalls a suite of protocols and procedures are proposed that can combat caller ID spoofing on VOIP and connected public telephone networks.
STIR/SHAKEN
Secure Telephony Identity Revisited / Signature-based Handling of Asserted information using toKENs
Used by robocallers to mask their identity or to make it appear the call is from a legitimate source usually orignates from voice-over-IP (VOIP) systems
STIR
Standards developed by the Internet Engineering Task Force (IETF)
For telecommunication service providers implement certificate management system to create and manage the public and private keys, digital certificates used to sign and verify Caller ID details.
Adds information to the SIP headers that allow the endpoints along the system to positively identify the origin of the data , such as JSON web tokens encrypted with the provider’s private key, encoded using Base64,
There are three levels of verification, or “attestation”
A : Full Attestation indicates that the provider recognizes the entire phone number as being registered with the originating subscriber.
B : Partial Attestation call originated with a known customer but the entire number cannot be verified,
C : Gateway Attestation call can only be verified as coming from a known gateway
How can the Public Key Infrastructure be used ?
In an interconnection network , each telephone service provider will obtain its digital certificate from a certificate authority (CA) that is trusted by other telephone service providers. Calling party signs the SIP Header caller ID as legitimate . The called party verifies that the calling number is authentic
STIR
Originating service provider’s encrypted SIP Identity Header includes the following data:
Attestation level
Date and Time
Calling and Called Numbers
Orig ID for analytics and/or traceback purposes among others
Location of certificate repository
Signature
Encryption algorithm
FCC has also assigned the role of a Secure Telephone Identity Policy Administrator (STI-PA) which oversees that CAs do not provide certificate to spoofing robocallers and enforce the framework for STIR /SHAKEN .
STIR is based on the SIP protocol and is designed to work with calls being routed through a VOIP network. Since traditional endpoints like POTS and SS7 networks also should be covered under this call authenticity framework , SHAKEN was developed to manage call via IP-to-telephone gateways .
Developed by the Alliance of Telecommunications Industry Solutions (ATIS)
Working Steps :
When a call is initiated, a SIP INVITE is received by the originating service provider.
Originating service provider verifies the call source and number to determine how to confirm validity.
Full Attestation (A) — The service provider authenticates the calling party AND confirms they are authorized to use this number. An example would be a registered subscriber.
Partial Attestation (B) — The service provider verifies the call origination but cannot confirm that the call source is authorized to use the calling number. An example would be a calling number from behind an enterprise PBX.
Gateway Attestation (C) — The service provider authenticates the call’s origin but cannot verify the source. An example would be a call received from an international gateway.
Create a SIP Identity header that contains information on the calling number, called number, attestation level, and call origination, along with the certificate thus caller ID “signed” as legitimate
SIP INVITE with the SIP Identity header with the certificate is sent to the destination service provider.
Destination service provider verifies the identity of the header and certificate.
Diagrammatic depiction of flow of how Telecom carriers to digitally validates authenticity before receiving or handoff through their network
SHAKEN
References
RFC 8224 – Authenticated Identity Management in the Session Initiation Protocol (SIP)
RFC 8226 – Secure Telephone Identity Credentials: CertificatesRFC 8588 – Personal Assertion Token (PaSSporT) Extension for Signature-based Handling of Asserted information using toKENs (SHAKEN)
SBC ( Session Borde Controllers ) are basically gateways that provide interconnectivity between the hosted IP-PBX of the enterprise to the outside world endpoints such as telco service provider, PSTN/ TDM , SIP trunking providers or even third party OTT provider apps like skype for business etc.
If you have a hosted IPPBX or PBX in your data-centre or on premise and you need controlled but heavy outflowing traffic, it is a good idea to integrate a resilient and efficient SBC to provide seamless interconnectivity.
Hosted PBX
For an enterprises such as an Trading floor or warehouse with multiple phone types , softphones , hardphones , turrets etc distributed across various geographies and zones a device agnostic architectural setup is prime . Listing the essentials for setting up such a system. Note supplementary services are data-services , logging , licensing etc are important but kept out of scope to keep focus on functional aspects .
An enterprise application usually is structured in tiers or layers
Client tier – the networks clients communication to the central java programs . Runs on client machines
web tier – state full communication between client and business tier . Runs in server machine.
business tier- handles the logic of the application. The business tier uses the Enterprise Java Bean (EJB) container, which manages the execution of the beans
data tier – encompasses DB drivers . Runs on separate machines for database storage
Event services for Line status notifications
providers lines status notification across enterprise for inter zone and softphone to hardphone .
Routing services
routing calls within enterprise and hardphone sites read more about resource zones later in the article
Call Control Manager (CCM)
consolidated set of all service and component that make up the VOIP platform besides media handlers . It includes SIP adapters , bridge managers , call processing frameworks , API frameworks , healthchecks etc .
Call processing framework ( CPF)
signalling and call routing logic , mostly in SIP and trunks . Manages identities such as Call Line information , Called Party Information , line status etc in shared memory.
Multiple shared Lines and their statuses
Incases where there is a need to process multiple calls from a single User agent device such as a softphone or hardphone ( common scenario for a turret phone) , the design involves assigning it multiple sip uris and each sip uri will establish a line.
When caller calls callee , the line is said to be BUSY , otherwise said to be IDLE. Transition of a shared sip line from IDLE to BUSY is transmitted to others via SIP PUBLISH as other UAs holding the same sip
Similarly any other event like transfer is propagated to other via SIP UPDATE
Clustering Call control managers (CCM)
A Call Communication manager (CCM) from various zones should be able to cowork on call and session management and advanced features such as routing from home guest zone to home zone , call transfer , refer , barge etc. Designing a clustered setup will also provide elasticity , fail-over and high availability. Can use clustered , HA compliant framework such as Oracle Communication Application Server , suited for enterprise level deployments.
Call Replication and distributed memory management
A node will store two types of data: active sessions and passive sessions. The active sessions are used by the node and stored in cache. The passive sessions are the replicas from the other nodes’ active sessions. The passives sessions are stored on a persistent storage.
Controlling Line Calls using AOR and Resource Zones
When dealing with many SIP endpoints , now referred to as resource, it is best to assign the resources to their respective zones. Thus a resource’s status updates will be only updated by its active resource zone while can be read by any resource zone.
Incoming request Zone vs Active Resource Zone
For an Incoming request such a INVITE , check whether the zone sending the request is its active resource zone or not .If the Active Resource Zone is the same zone on which the INVITE came in, then the call is handled by that zone. If the Active Resource Zone is a different zone, then the call needs to be forwarded to the Active Resource Zone.
Bridges for Local Media connections
Although call signalling is handled by a resources active resource zone only, we can still create media bridges in local zone of the resource .
Local MM bridges are used to auto answer an incoming sip line call and create trunk , especially from hardphones which do not support provisional responses.
Interzone proxy Handler
proxies call control messages between active and non active resource zones. Primarily mapping the sip messages with all custom headers inbetween the communication device interfaces.
Dial Trunk using multiple dedicated sip lines and connect via Media Bridge
To save up on call routing /connection time and to support te ability to add as many users on call at runtime , a dedicated media bridge is established for every call.
A sip line activated is auto-answered by MM , creates a trunk and waits for other endpoint to join the bridge. The flow is as follows :
As INVITE arrives for an IDLE sip line , it is connected to a trunk and auto answered by a local MM bridge .
Since the call is already answered , when caller dials number for callee , collect the DTMF digits over RTP using RFC 2833 DTMF events.
Run inter-digit timer for digit collection and detect end of dialing on timeout.
The dialed trunk connection is made and call is added to media bridge
When provisional responses are received on the trunk connection, generate in-band call progress tones (ringing, proceeding etc) via the MM
When the line answers, the progress tones have to be stopped and the called party gets bridged to the calling party via the media bridge.
Call Diversion involves forwarding calls from zone to another zone. joinjed parties get call UPDATE status and forward response .
Call barge is the processing of joining an ongoing call . The barge event is usually propagated to joined parities via SIP INFO. Private lines do not allow barge in and are exclusively reserved for only few users.
Interconnectivity provided by an SBC ( Session Border Controller)
Hold-Resume and Music on Hold in multi-line evironment
While a regular p2p call involves simple reinvite based hold and resume with varrying SDP, the scenario is slightly more detailed for hold resume on bridged trunk connection , as explained below.
As the calls made are on bridge , a hold signal involves a RE-INIVITE with held-SDP to media manager (MM). If hold status on trunk is 200 OK the hold status will be sent to other call interfaces connected on the trunk. Else if hold is denied ,403 is sent back to hold-initiates.
Music on hold is an one way RTP mostly from media server.
For a bridged scenarios , separate Music on hold bridges are kept on Media Managers. When an UA has to hold , it is removed from original bridge and place on music on hold bridge . To be unhold/ resume it is placed back into the orignal bridge from music on hold bridge .
Conference
user initiates conference, the conference feature can execute on the zone where the user was logged on, irrespective of zones where the other conference attendees join from . The Call processing framework of originators zone completes the SDP exchange to establish two-way speech path among all the parties.
Incases there are multiple connections from a zone , a local MM conference bridge can be created for them which would connect back to originators MM conf bridge . this two part conf bridge will be transparent to the sip line sand users .
For provisioning inputs and settings setup a Diagnostics , Administration and Configuration platform which can process APIs for data services , licences , alarms or do remote device control such as using SNMP
Session Border Controllers (SBC)
At network level SBC operations include
bridging multiple interfaces in different networks even between the IPv4 and IPv6 networks
auto NAT discovery and STUN
protocol conversion such as TLS to UDP etc
Flood detection and IP filtering
For SIP specific functionalities , SBC does
SIP validation involving checks on syntax and message contents also consistency checks are performed.
stateful and call aware. tracing, monitoring and checking for validitya and health of all the SIP messages
Topology hiding
Traffic filtering
Codec filtering , reordering , media pinning, transcoding, or call recording
Data replication brings High Availability (HA) with hot backups or even Active-Active solutions.
Traffic sharing and routing roles of SBC can include
IP-based and Digest-based authentication
limiting traffic by number of concurrent calls or calling rate.
Dialplan and/or Custom routing
Dispatching/Load-balancing to a backend cluster of servers
SBC’s can be physical hardware boxes or software based applications, as the name suggests their purpose is to control the session at border between the enterprise and external service provider.
SIP to PSTN – SIP is an IP protocol whereas PSTN is a TDM one , achieving interoperability is also the KRA of an SBC
SIP trunking – SBC provide a secure sip connectivity to connect calls to sip trunks which provide bulk calls functionality at a flat pricing.
support for various fixed or mobile endpoints – SBC ensure they are RFC compliant and can extend SIP to any kind of telecom endpoint like PSTN , GSM, fax , Skype , sipphone , IP phones etc.
NAT / Network address translator – To meet the packet routing challenges across a firewall or even during private -public mapping. A combo of DHCP servers and NAT provider comes very handy to reroute or perform hole punching such that signalling and media packets are not dropped and meet the required endpoint. More about NAT here – NAT traversal using STUN and TURN.
Load balancing – Reverse proxies and Load balancers is a much adopted industry practise to mask the inner IPs of the VoIP platform and also route traffic appropriately between control and media server .
Security , QoS and Regulatory compliance – since SBCs are required to typically support a large array of clients they adhere to regulatory and industry accepted standards ,which also involves security features like AAA, TLS/SSL and other means for quality of assurance like logging and fault detection, preventing DDoS etc . In many cases SBC can also encrypt / decrypt RTP streams for probing , tapping or lawful inspection .
Terminating at carriers , PSTN and IP gateways
Additional SBC features
Inaddition to above it is good to have if an SBC provides extra features like forking , emergency number dialing ( 911 ) or active directory integration . Real Time Analysis and monitoring of call and metrics are also expected from a SBC since they reside on edge of the network and are more vulnerable to threats . For example Dialogic Mediant SBC’s and gateways , Audio Codes SBCs
With the shift from on premise PBXs to cloud based VM or microservice architecture , SBC vendors adopt a lager umbrella of services also including automation scripts for checks , reporting tools / consoles , developer friendly APIs to manage sessions via SBC and even WebRTC gateways to connect browser endpoints .
Usage Scenarios
Any VOIP dependant system which deals with bulksome voice / video traffic from external endpoints is a usages scenarios. Listing few
Contact Call centres
Remote work / offsite monitoring
CRM solution for sales/marketing
Connecting webrtc click to dial from webpage to enterprise representatives
connecting enterprise UCC clients to PSTN endpoints
Continuous Integration and Delivery Automation using Jenkins
continous delivery hub
distribute work across multiple machines, helping drive builds, tests and deployments across multiple platforms
self-contained Java-based program
extensible using plugins
Jenkins pieline
orchestrate and automate building their project in Jenkins
Configuration management using chef cookbooks
Alternatives like puppet and Ansible, which are also a cross-platform configuration management platform
Compute virtualization and containerization using Docker
Docker containers can be used instead of virtual machines such as VirtualBox , to isolates applications and be OS and platform independent
Makes distributed development possible and automates the deployment possible
Usage: docker [OPTIONS] COMMAND
A self-sufficient runtime for containers
Options:
–config string Location of client config files (default “/root/.docker”)
-D, –debug Enable debug mode
-H, –host list Daemon socket(s) to connect to
-l, –log-level string Set the logging level (“debug”|”info”|”warn”|”error”|”fatal”) (default “info”)
–tls Use TLS; implied by –tlsverify
–tlscacert string Trust certs signed only by this CA (default “/root/.docker/ca.pem”)
–tlscert string Path to TLS certificate file (default “/root/.docker/cert.pem”)
–tlskey string Path to TLS key file (default “/root/.docker/key.pem”)
–tlsverify Use TLS and verify the remote
-v, –version Print version information and quit
Docker commands
Management Commands:
builder Manage builds
config Manage Docker configs
container Manage containers
engine Manage the docker engine
image Manage images
network Manage networks
node Manage Swarm nodes
plugin Manage plugins
secret Manage Docker secrets
service Manage services
stack Manage Docker stacks
swarm Manage Swarm
system Manage Docker
trust Manage trust on Docker images
volume Manage volumes
Commands:
attach Attach local standard input, output, and error streams to a running container
build Build an image from a Dockerfile
commit Create a new image from a container’s changes
cp Copy files/folders between a container and the local filesystem
create Create a new container
diff Inspect changes to files or directories on a container’s filesystem
events Get real time events from the server
exec Run a command in a running container
export Export a container’s filesystem as a tar archive
history Show the history of an image
images List images
import Import the contents from a tarball to create a filesystem image
info Display system-wide information
inspect Return low-level information on Docker objects
kill Kill one or more running containers
load Load an image from a tar archive or STDIN
login Log in to a Docker registry
logout Log out from a Docker registry
logs Fetch the logs of a container
pause Pause all processes within one or more containers
port List port mappings or a specific mapping for the container
ps List containers
pull Pull an image or a repository from a registry
push Push an image or a repository to a registry
rename Rename a container
restart Restart one or more containers
rm Remove one or more containers
rmi Remove one or more images
run Run a command in a new container
save Save one or more images to a tar archive (streamed to STDOUT by default)
search Search the Docker Hub for images
start Start one or more stopped containers
stats Display a live stream of container(s) resource usage statistics
stop Stop one or more running containers
tag Create a tag TARGET_IMAGE that refers to SOURCE_IMAGE
top Display the running processes of a container
unpause Unpause all processes within one or more containers
update Update configuration of one or more containers
version Show the Docker version information
wait Block until one or more containers stop, then print their exit codes
docker container commands
List all container
docker ps
docker container COMMAND
attach Attach local standard input, output, and error streams to a running container
commit Create a new image from a container’s changes
cp Copy files/folders between a container and the local filesystem
create Create a new container
diff Inspect changes to files or directories on a container’s filesystem
exec Run a command in a running container
export Export a container’s filesystem as a tar archive
inspect Display detailed information on one or more containers
kill Kill one or more running containers
logs Fetch the logs of a container
ls List containers
pause Pause all processes within one or more containers
port List port mappings or a specific mapping for the container
prune Remove all stopped containers
rename Rename a container
restart Restart one or more containers
rm Remove one or more containers
run Run a command in a new container
start Start one or more stopped containers
stats Display a live stream of container(s) resource usage statistics
stop Stop one or more running containers
top Display the running processes of a container
unpause Unpause all processes within one or more containers
update Update configuration of one or more containers
wait Block until one or more containers stop, then print their exit codes
docker image commands
see all iamges
>docker images
REPOSITORY TAG IMAGE ID CREATED SIZE
sipcapture/homer-cron latest fb2243f90cde 3 hours ago 476MB
sipcapture/homer-kamailio latest f159d46a22f3 3 hours ago 338MB
sipcapture/heplify latest 9f5280306809 21 hours ago 9.61MB
<none> <none> edaa5c708b3a 21 hours ago 619MB
mysql 5.6 c30095c52827 36 hours ago 256MB
sipcapture/homer-app master 1e883a8d88d4 2 days ago 454MB
postgres 11-alpine 5239fade3a90 7 days ago 71.9MB
golang alpine 6b21b4c6e7a3 7 days ago 350MB
alpine latest b7b28af77ffe 7 days ago 5.58MB
debian jessie 652b7a59e393 9 days ago 129MB
sipcapture/heplify-server master aa85bfa7cb3e 2 weeks ago 22.6MB
kapacitor 1.5-alpine 2a63b9d348df 4 weeks ago 73.6MB
influxdb 1.5-alpine 40e13f6ee02a 7 weeks ago 84MB
chronograf 1.5-alpine 02ed2863e25b 7 weeks ago 44.5MB
sipcapture/hepsub master 3ef6550c4bc6 2 months ago 90.6MB
hello-world latest fce289e99eb9 6 months ago 1.84kB
sipcapture/homer-webapp latest 75e5b5b7b33c 7 months ago 428MB
telegraf 1.5-alpine aa8daabb3b1c 10 months ago 42MB
stefanprodan/caddy latest 655880563633 21 months ago 24.7MB
See all stats
>docker stats
CONTAINER ID NAME CPU % MEM USAGE / LIMIT MEM % NET I/O BLOCK I/O PIDS
f42c71741107 homer-cron 0.00% 52KiB / 994.6MiB 0.01% 2.3kB / 0B 602MB / 0B 0
0111765091ae mysql 0.04% 452.2MiB / 994.6MiB 45.46% 1.35kB / 0B 2.06GB / 49.2kB 22
Run command from with docker
docker exec -it bash
First see all processes
docker ps
select a process and enter its bash
docker exec -it 0472a5127fff bash
to edit or update a file inside docker either install vim everytime u login in resh docker conainer like
apt-get update
apt-get install vim
or add this to dockerfile
RUN [“apt-get”, “update”]
RUN [“apt-get”, “install”, “-y”, “vim”]
see if ngrep is install , if not then install and run ngrep to get sip logs isnode that docker container
apt update
apt install ngrep
ngrep -p "14795778704" -W byline -d any port 5060
docker volume
volumes are used for persisting data generated by and used by Docker containers.
docker volumes have advantages over blind mounts such as
easier to backup or migrate , managed by docker APIs, can be safely shared among multiple containers etc
Commands:
create Create a volume
inspect Display detailed information on one or more volumes
ls List volumes
prune Remove all unused local volumes
rm Remove one or more volumes
docker stack
Lets to manager a cluster of docker containers thorugh docker swarm
can be defined via docker-compose.yml file
Options:
–orchestrator string Orchestrator to use (swarm|kubernetes|all)
Commands:
deploy Deploy a new stack or update an existing stack
ls List stacks
ps List the tasks in the stack
rm Remove one or more stacks
services List the services in the stack
docker service
commands :
create Create a new service
inspect Display detailed information on one or more services
logs Fetch the logs of a service or task
ls List services
ps List the tasks of one or more services
rm Remove one or more services
rollback Revert changes to a service’s configuration
scale Scale one or multiple replicated services
update Update a service
Run docker ccontainers
Run a command in a new container
Options:
–add-host list Add a custom host-to-IP mapping (host:ip)
-a, –attach list Attach to STDIN, STDOUT or STDERR
–blkio-weight uint16 Block IO (relative weight), between 10 and 1000, or 0 to disable (default 0)
–blkio-weight-device list Block IO weight (relative device weight) (default [])
–cap-add list Add Linux capabilities
–cap-drop list Drop Linux capabilities
–cgroup-parent string Optional parent cgroup for the container
–cidfile string Write the container ID to the file
–cpu-period int Limit CPU CFS (Completely Fair Scheduler) period
–cpu-quota int Limit CPU CFS (Completely Fair Scheduler) quota
–cpu-rt-period int Limit CPU real-time period in microseconds
–cpu-rt-runtime int Limit CPU real-time runtime in microseconds
-c, –cpu-shares int CPU shares (relative weight)
–cpus decimal Number of CPUs
–cpuset-cpus string CPUs in which to allow execution (0-3, 0,1)
–cpuset-mems string MEMs in which to allow execution (0-3, 0,1)
-d, –detach Run container in background and print container ID
–detach-keys string Override the key sequence for detaching a container
–device list Add a host device to the container
–device-cgroup-rule list Add a rule to the cgroup allowed devices list
–device-read-bps list Limit read rate (bytes per second) from a device (default [])
–device-read-iops list Limit read rate (IO per second) from a device (default [])
–device-write-bps list Limit write rate (bytes per second) to a device (default [])
–device-write-iops list Limit write rate (IO per second) to a device (default [])
–disable-content-trust Skip image verification (default true)
–dns list Set custom DNS servers
–dns-option list Set DNS options
–dns-search list Set custom DNS search domains
–entrypoint string Overwrite the default ENTRYPOINT of the image
-e, –env list Set environment variables
–env-file list Read in a file of environment variables
–expose list Expose a port or a range of ports
–group-add list Add additional groups to join
–health-cmd string Command to run to check health
–health-interval duration Time between running the check (ms|s|m|h) (default 0s)
–health-retries int Consecutive failures needed to report unhealthy
–health-start-period duration Start period for the container to initialize before starting health-retries countdown (ms|s|m|h) (default 0s)
–health-timeout duration Maximum time to allow one check to run (ms|s|m|h) (default 0s)
–help Print usage
-h, –hostname string Container host name
–init Run an init inside the container that forwards signals and reaps processes
-i, –interactive Keep STDIN open even if not attached
–ip string IPv4 address (e.g., 172.30.100.104)
–ip6 string IPv6 address (e.g., 2001:db8::33)
–ipc string IPC mode to use
–isolation string Container isolation technology
–kernel-memory bytes Kernel memory limit
-l, –label list Set meta data on a container
–label-file list Read in a line delimited file of labels
–link list Add link to another container
–link-local-ip list Container IPv4/IPv6 link-local addresses
–log-driver string Logging driver for the container
–log-opt list Log driver options
–mac-address string Container MAC address (e.g., 92:d0:c6:0a:29:33)
-m, –memory bytes Memory limit
–memory-reservation bytes Memory soft limit
–memory-swap bytes Swap limit equal to memory plus swap: ‘-1’ to enable unlimited swap
–memory-swappiness int Tune container memory swappiness (0 to 100) (default -1)
–mount mount Attach a filesystem mount to the container
–name string Assign a name to the container
–network string Connect a container to a network (default “default”)
–network-alias list Add network-scoped alias for the container
–no-healthcheck Disable any container-specified HEALTHCHECK
–oom-kill-disable Disable OOM Killer
–oom-score-adj int Tune host’s OOM preferences (-1000 to 1000)
–pid string PID namespace to use
–pids-limit int Tune container pids limit (set -1 for unlimited)
–privileged Give extended privileges to this container
-p, –publish list Publish a container’s port(s) to the host
-P, –publish-all Publish all exposed ports to random ports
–read-only Mount the container’s root filesystem as read only
–restart string Restart policy to apply when a container exits (default “no”)
–rm Automatically remove the container when it exits
–runtime string Runtime to use for this container
–security-opt list Security Options
–shm-size bytes Size of /dev/shm
–sig-proxy Proxy received signals to the process (default true)
–stop-signal string Signal to stop a container (default “SIGTERM”)
–stop-timeout int Timeout (in seconds) to stop a container
–storage-opt list Storage driver options for the container
–sysctl map Sysctl options (default map[])
–tmpfs list Mount a tmpfs directory
-t, –tty Allocate a pseudo-TTY
–ulimit ulimit Ulimit options (default [])
-u, –user string Username or UID (format: [:])
–userns string User namespace to use
–uts string UTS namespace to use
-v, –volume list Bind mount a volume
–volume-driver string Optional volume driver for the container
–volumes-from list Mount volumes from the specified container(s)
-w, –workdir string Working directory inside the container
sample run command
docker run -it -d --name opensips -e ENV=dev imagename:2.2
-it flags attaches to an interactive tty in the container.
-e gives envrionment variables
-d runs it in background and prints container id
remove docker entities
To remove all stopped containers, all dangling images, and all unused networks:
docker system prune -a
To remove all unused volumes
docker system prune --volumes
To remove all stopped containers
docker container prune
sometimes docker images keep piling with stopped congainer such as
REPOSITORY TAG IMAGE ID CREATED SIZE d1dcfe2438ae 15 minutes ago 753MB 2d353828889b 16 hours ago 910MB fb16e1be51f3 16 hours ago 910MB 3dc9b3fbe2fb 17 hours ago 910MB 7d1e4f183d93 18 hours ago 910MB 70fe962b9971 18 hours ago 910MB c349859656b5 18 hours ago 910MB 5d4cbd1f4cbe 18 hours ago 910MB aef6e40820f2 18 hours ago 910MB 4a4d3897f40e 18 hours ago 910MB 46bca8b4b1c3 20 hours ago 910MB c172e5f24798 20 hours ago 910MB
CONTAINER ID IMAGE COMMAND CREATED STATUS PORTS NAMES
0dd6698a7517 2d353828889b "/entrypoint.sh" 13 minutes ago Exited (137) 13 minutes ago hardcore_wozniak
047a42f7a6cd d1dcfe2438ae "/bin/sh -c 'aws s3 …" 18 minutes ago Exited (1) 18 minutes ago ecstatic_heisenberg
65b2305520e9 2d353828889b "/entrypoint.sh" 27 minutes ago Exited (137) 26 minutes ago mystifying_robinson
56841a6c8da4 2d353828889b "/entrypoint.sh" 17 hours ago Exited (137) 26 minutes ago compassionate_keldysh
c23f9399c53d zt-voipmonitor "/entrypoint.sh" 17 hours ago Exited (7) 17 hours ago angry_zhukovsky
5a0f99b7485a 3dc9b3fbe2fb "/entrypoint.sh" 17 hours ago Exited (137) 17 hours ago elegant_haslett
f49a4305928a 7d1e4f183d93 "/entrypoint.sh" 17 hours ago Exited (137) 17 hours ago boring_bose
53afb5633a60 7d1e4f183d93 "/entrypoint.sh" 17 hours ago Exited (137) 17 hours ago suspicious_varahamihira
662f6a76ecb3 7d1e4f183d93 "/entrypoint.sh" 17 hours ago Exited (137) 17 hours ago wizardly_booth
c5dd4e8e8a12 7d1e4f183d93 "/entrypoint.sh" 17 hours ago Exited (137) 17 hours ago gracious_kapitsa
0d17dd07af57 7d1e4f183d93 "/entrypoint.sh" 17 hours ago Exited (137) 17 hours ago upbeat_elbakyan
60b419a51099 7d1e4f183d93 "/entrypoint.sh" 17 hours ago Exited (137) 17 hours ago dreamy_ptolemy
6f3a7ca0a2d4 7d1e4f183d93 "/entrypoint.sh" 17 hours ago Exited (137) 17 hours ago stupefied_gauss
0acab30d0c33 7d1e4f183d93 "/entrypoint.sh" 17 hours ago Exited (137) 17 hours ago reverent_varahamihira
b3414b91d718 7d1e4f183d93 "/entrypoint.sh" 18 hours ago Exited (137) 17 hours ago blissful_goodall
to remove such images and their conainer , first stop and remove confainers
docker stop $(docker ps -a -q)
docker rm $(docker ps -a -q)
building, changing, and versioning infrastructure Infra as Code – can run single application to datacentres via configuration files which create execution plan can manage low-level components such as compute instances, storage, and networking, as well as high-level components such as DNS entries, SaaS features, etc. Resource Graph – builds a graph of all your resources
tfenv can be used to manage terraform versions
brew unlink terraform
tfenv install 0.11.14
tfenv list
Terraform configuration language
used for decalring resoirces and descriptions of infrastructure
.tf or .tf.json file extension
group of resources can be gathered into a module
Terraform configuration consists of a root module, where evaluation begins, along with a tree of child modules created when one module calls another.
Exmaple : launch a single AWS EC2 instance , fle server1.tf
console Interactive console for Terraform interpolations
destroy Destroy Terraform-managed infrastructure
env Workspace management
fmt Rewrites config files to canonical format
get Download and install modules for the configuration
graph Create a visual graph of Terraform resources
import Import existing infrastructure into Terraform
init Initialize a Terraform working directory
output Read an output from a state file
plan Generate and show an execution plan
providers Prints a tree of the providers used in the configuration
refresh Update local state file against real resources
show Inspect Terraform state or plan
taint Manually mark a resource for recreation
untaint Manually unmark a resource as tainted
validate Validates the Terraform files
version Prints the Terraform version
workspace Workspace management
0.12upgrade Rewrites pre-0.12 module source code for v0.12
debug Debug output management (experimental)
force-unlock Manually unlock the terraform state
push Obsolete command for Terraform Enterprise legacy (v1)
state Advanced state management
terraform init
initialize a working directory containing Terraform configuration files.
terraform validate
checks that verify whether a configuration is internally-consistent, regardless of any provided variables or existing state.
Kubernetes
container orchestration platform , automating deployment, scaling, and management of containerized applications. Can deploy to cluster of computers, automating the distribution and scheduling as well
Service discovery and load balancing –
gives Pods their own IP addresses and a single DNS name for a set of Pods, and can load-balance across them.
Automatic bin packing –
Automatically places containers based on their resource requirements and other constraints, while not sacrificing availability. Mix critical and best-effort workloads in order to drive up utilization and save even more resources.
Storage orchestration –
Automatically mount the storage system of your choice, whether from local storage, a public cloud provider such as GCP or AWS, or a network storage system such as NFS, iSCSI, Gluster, Ceph, Cinder, or Flocker.
Self-healing –
Restarts containers that fail, replaces and reschedules containers when nodes die, kills containers that don’t respond to your user-defined health check, and doesn’t advertise them to clients until they are ready to serve.
Automated rollouts and rollbacks –
progressively rolls out changes to your application or its configuration, while monitoring application health to ensure it doesn’t kill all your instances at the same time.
Secret and configuration management –
Deploy and update secrets and application configuration without rebuilding your image and without exposing secrets in your stack configuration.
Batch execution-
manage batch and CI workloads, replacing containers that fail, if desired.
Horizontal scaling –
Scale application up and down with a simple command, with a UI, or automatically based on CPU usage.
Starting Kubernetes…minikube version: v1.3.0
commit: 43969594266d77b555a207b0f3e9b3fa1dc92b1f
minikube v1.3.0 on Ubuntu 18.04
Running on localhost (CPUs=2, Memory=2461MB, Disk=47990MB) …
OS release is Ubuntu 18.04.2 LTS
Preparing Kubernetes v1.15.0 on Docker 18.09.5 …
kubelet.resolv-conf=/run/systemd/resolve/resolv.conf
Pulling images …
Launching Kubernetes …
Done! kubectl is now configured to use "minikube"
dashboard was successfully enabled
Kubernetes Started
Basic Commands
start Starts a local kubernetes cluster
status Gets the status of a local kubernetes cluster
stop Stops a running local kubernetes cluster
delete Deletes a local kubernetes cluster
dashboard Access the kubernetes dashboard running within the minikube cluster
Images Commands:
docker-env Sets up docker env variables; similar to ‘$(docker-machine env)’
cache Add or delete an image from the local cache.
Configuration and Management Commands:
addons Modify minikube’s kubernetes addons
config Modify minikube config
profile Profile gets or sets the current minikube profile
update-context Verify the IP address of the running cluster in kubeconfig.
Networking and Connectivity Commands:
service Gets the kubernetes URL(s) for the specified service in your local cluster
tunnel tunnel makes services of type LoadBalancer accessible on localhost
Advanced Commands:
mount Mounts the specified directory into minikube
ssh Log into or run a command on a machine with SSH; similar to ‘docker-machine ssh’
kubectl Run kubectl
Troubleshooting Commands:
ssh-key Retrieve the ssh identity key path of the specified cluster
ip Retrieves the IP address of the running cluster
logs Gets the logs of the running instance, used for debugging minikube, not user code.
update-check Print current and latest version number
version Print the version of minikube
Other Commands:
completion Outputs minikube shell completion for the given shell (bash or zsh)
kubectl
controls the Kubernetes cluster manager.
Basic Commands (Beginner):
create Create a resource from a file or from stdin.
expose Take a replication controller, service, deployment or pod and expose it as a new Kubernetes Service
run Run a particular image on the cluster
set Set specific features on objects
explain Documentation of resources
get Display one or many resources
edit Edit a resource on the server
delete Delete resources by filenames, stdin, resources and names, or by resources and label selector
Deploy Commands:
rollout Manage the rollout of a resource
scale Set a new size for a Deployment, ReplicaSet, Replication Controller, or Job
autoscale Auto-scale a Deployment, ReplicaSet, or ReplicationController
Cluster Management Commands:
certificate Modify certificate resources.
cluster-info Display cluster info
top Display Resource (CPU/Memory/Storage) usage.
cordon Mark node as unschedulable
uncordon Mark node as schedulable
drain Drain node in preparation for maintenance
taint Update the taints on one or more nodes
Troubleshooting and Debugging Commands:
describe Show details of a specific resource or group of resources
logs Print the logs for a container in a pod
attach Attach to a running container
exec Execute a command in a container
port-forward Forward one or more local ports to a pod
proxy Run a proxy to the Kubernetes API server
cp Copy files and directories to and from containers.
auth Inspect authorization
Advanced Commands:
diff Diff live version against would-be applied version
apply Apply a configuration to a resource by filename or stdin
patch Update field(s) of a resource using strategic merge patch
replace Replace a resource by filename or stdin
wait Experimental: Wait for a specific condition on one or many resources.
convert Convert config files between different API versions
kustomize Build a kustomization target from a directory or a remote url.
Settings Commands:
label Update the labels on a resource
annotate Update the annotations on a resource
completion Output shell completion code for the specified shell (bash or zsh)
Other Commands:
api-resources Print the supported API resources on the server
api-versions Print the supported API versions on the server, in the form of “group/version”
config Modify kubeconfig files
plugin Provides utilities for interacting with plugins.
version Print the client and server version information
DevOps monitoring tools nagios
Manage Docker configs
Commands:
create Create a config from a file or STDIN
inspect Display detailed information on one or more configs
ls List configs
rm Remove one or more configs
Manage containers
Commands:
attach Attach local standard input, output, and error streams to a running container
commit Create a new image from a container’s changes
cp Copy files/folders between a container and the local filesystem
create Create a new container
diff Inspect changes to files or directories on a container’s filesystem
exec Run a command in a running container
export Export a container’s filesystem as a tar archive
inspect Display detailed information on one or more containers
kill Kill one or more running containers
logs Fetch the logs of a container
ls List containers
pause Pause all processes within one or more containers
port List port mappings or a specific mapping for the container
prune Remove all stopped containers
rename Rename a container
restart Restart one or more containers
rm Remove one or more containers
run Run a command in a new container
start Start one or more stopped containers
stats Display a live stream of container(s) resource usage statistics
stop Stop one or more running containers
top Display the running processes of a container
unpause Unpause all processes within one or more containers
update Update configuration of one or more containers
wait Block until one or more containers stop, then print their exit codes
Alternatives, Senu multi-cloud monitoring or Raygun
Metrics for monitoring a VOIP call can be obtained from any node in media path of the call flow . Essentially used for analysis via calculation and aggregation , and sometimes used for realtime performance tracking and rectification too .
Rating Factor (R-Factor) and Mean Opinion Score (MOS) are two commonly-used measurements of overall VoIP call quality.
R-Factor: A value derived from metrics such as latency, jitter, and packet loss per ITU‑T Recommendation G.107. It assess the quality-of-experience for VoIP calls on your network. Typical scores range from 50 (bad) to 90 (excellent).
R factor of 90 , Mos is 4.3 ( Excellent )
R factor 50 , Mos is 2.6 ( Bad)
MOS: It is derived from the R-Factor per ITU‑T Recommendation G.10 which measures VoIP call quality. PacketShaper measures MOS using a scale of 10-50. To convert to a standard MOS score (which uses a scale of 1-5), divide the PacketShaper MOS value by 10.
ITU ? The International Telecommunication Union is the United Nations specialised agency in the field of telecommunications, information and communication technologies (ICTs).
ITU-T ? TU Telecommunication Standardisation Sector is responsible for studying technical, operating and tariff questions and issuing Recommendations on them with a view to standardising telecommunications on a worldwide basis.
MOS is terminology for audio, video and audiovisual quality expressions as per ITU-T P.800.1. It refers to listening, talking or conversational quality, whether they originate from subjective or objective models.
Very Good: 4.3-5.0
Bad: 3.1-3.6
Not Recommenced : 2.6-3.1
Very Bad: 1.0-2.6
It provides provisions for identifiers regarding the audio bandwidth, the type of interface (electrical or acoustical) and the video resolution too , such as
MOS-AVQE for audiovisual quality;
MOS-CQE is for estimated conversational quality;
MOS-LQE for listening quality;
MOS-TQE is used for talking quality;
MOS-VQE depicts video quality;
For Audio Signal Speech Quality/ AV
– N denotes audio signals upto narrow-band (300-3400 Hz)
– W is for audio signals upto wideband (50-7000 Hz)
– S for upto super-wideband (20-14000 Hz)
– F is obtained for fullband (10-20000 Hz)
For Listening quality LQO
electrical measurement
performed at electrical interfaces only. In order to predict the listening quality as perceived by the user, assumptions for the terminals are made in terms of intermediate reference system (IRS) or corrected IRS frequency response. A sealed condition between the handset receiver and the user’s ear is assumed.
acoustical measurement
performed at acoustical interfaces. In order to predict the listening quality as perceived by the user, this measurement includes the actual telephone set products provided by the manufacturer or vendor. In combination with the choice of the acoustical receiver in the laboratory test , there will be a more or less leaky condition between the handset’s receiver and the artificial ear.
Conversational Quality / CQ
Arithmetic mean value of subjective judgments on a 5-point ACR quality scale, is calculated. Talking Quality / TQ
This describes the quality of a telephone call as it is perceived by the talking party only. Factors affecting TQ include echo signal , background noise , double talk etc. It is calculated based on the arithmetic mean value of judgments on a 5-point ACR quality scale.
Video Quality / VQ
To account for differentiation in perceived quality for mobile and fixed devices and to allow for proper handling of different use-cases as
– M for mobile screen such as a smartphone or tablet (approximately 25 cm or less)
– T for PC/TV monitors
It is calculated based on the arithmetic mean value of subjective judgments, typically on a 5-point quality scale
Audio Visual Quality / AVQ
Refers to quality of audio visual stream under corresponding networking conditions. It is also calculated based on the arithmetic mean value of judgments on a 5-point ACR quality scale.
Other parameters also contributing to VoIP metric Analysis
Latency
It is the time required for packets to travel from one end to another, in milliseconds. If the sum of measured latency is 800 ms and the number of latency samples is 20, then the average latency is 40 ms. Header of the RTP packets carry timestamps which later can also be used to calculate round-trip time.
Round Trip Time
time taken for data to travel to the target destination and back. It is calculated as when the packet was sent and when acknowledgment for it was received.
Measured in milliseconds (ms), high RTT indicates a poor network quality and would result in the audio lag issue.
RTT can represent full path network latency experienced by the packets and can do away with frequent ICMP ping/echo requests/probes to check network health .
They are used to calculates RTO ( Request transmission timeouts )in TCP transmission ie how much time the sender should wait before retrying to send an unacknowledged packet.
Packet Loss
When packet does not successfully make it to the destination. It could happen due to multiple reasons such as
network bandwidth unavailable or network congestion
overloading of the buffer such that they do not have enough space to queue the packets or high priority preferences
intentionally configuring ACL or firewalls to drop the packets or discarding packets above rate limit by internet service provider
CPU unable to cope up with high security networks encryption and decryption speed requirements
Low battery on device may cause cause underworking of devices and hence lead to packet loss
limitation on physical device like softphone , hardphone or bluetooth headsets or if the hardware is broken at router , switch or cabling
for bluetooth headsets distance range could also be problem for weak signals and consequently packets drops
network errors as shown under Simple Network Management Protocol (SNMP) issues like FCS Errors, Alignment Errors, Frame Too Longs, MAC Receive Errors, Symbol Errors, Collisions, Carrier Sense Errors, Outbound Errors, Outbound Discards, Inbound Discards, Inbound Errors, and Unknown Protocol errors.
radio frequency interference from high voltage systems or microwaves can also cause packet drop in wireless networks
such that the packet can either not arrive or arrive late and be dropped out by the codec . To the listener it would appear like chopped voice or complete dropout for moments .
Obtaining packet loss details
Packet loss percentage is performed as per RFC 3550 using RTP header sequence numbers. If packets are missing sequence the media stream monitors flags that as lost packet.
It can also be concluded from the difference between total packets and received packets from CDR
RTP-XR (RFC-3611) records report real-time drops
Jitter
The variation in the delay of received packets in a flow, measured by comparing the interval when RTP packets were sent to the interval at which they were received. For instance, if packet #1 and packet #2 leave 30 milliseconds apart and arrive 50 milliseconds apart, then the jitter is 20 milliseconds or if packets transmitted every 15ms and reach destination at every 15ms then there is no variability and the jitter is 0.
Causes jitter
Frame bigger than jitter buffer size
algorithms to back-of collision by introducing delays in packet transmission in half duplex interfaces
even small jitter can get exponentially worse on slow or congestion links
jitter can be introduced due to bottlenecks near router buffer, rerouting / parallel routes to the same destination, load-sharing, or route tables changing the path
Handling jitter :
Jitter below 30ms is manageable with the help of jitter buffers in codecs however above that the codec starts to drop the late arrived packets and cannot reassemble / splice up the packets for a smooth media stream effectively, hence causing media quality issues like clipped audio
detecting jitter:
looking at inter packet gap in the direction of RTP stream in wireshark
RTP-XR (RFC-3611 & RFC-7005) for real-time jitter buffer usage and drops.
Methods for objective and subjective assessment of speech and video quality.
Mapping R-value to calculate MOS
To map MOS from R value using above defined metrics , a standard formula is used. First the latency and jitter are added and defined value for computation time is also added , resulting in effective latency
R = R – (lostPackets * impact)
MOS = ( (R - 60) * (100 – R) * 0.000007R) + 0.035R + 1)
MOS on RTP engine Kamailio
Minimum edge Values
mos_min_pv minimum encountered MOS value for the call. range – 1.0 to 5.0.
mos_min_at_pv timestamp of when the minimum MOS value was encountered during the call
mos_min_packetloss_pv amount of packetloss in percent at the time the minimum MOS value was encountered
mos_min_roundtrip_pv packet round-trip time in milliseconds at the time the minimum MOS value was encountered
mos_min_jitter_pv amount of jitter in milliseconds at the time the minimum MOS value was encountered
Maximum edge Values
mos_max_pv maximum encountered MOS value for the call.
mos_max_at_pv timestamp of when the maximum MOS value was encountered during the cal
mos_max_packetloss_pv amount of packetloss in percent at maximum MOS moment
mos_max_roundtrip_pv packet round-trip time in milliseconds at maximum MOS moment
mos_max_jitter_pv amount of jitter in milliseconds at maximum moment
Average Values
mos_average_pv average (median) MOS value for the call. Range – 1.0 through 5.0.
mos_average_packetloss_pv average (median) amount of packetloss in percent present throughout the call.
mos_average_jitter_pv average (median) amount of jitter in milliseconds present throughout the call.
mos_average_roundtrip_pv
mos_average_samples_pv number of samples used to determine the other “average” MOS data points.
Labels
mos_A_label_pv custom label used in rtpengine signalling. If set, all the statistics pseudovariables with the A suffix will be filled in with statistics only from the call legs that match the label given in this variable.
A label’s min mos_min_A_pv mos_min_at_A_pv mos_min_packetloss_A_pv mos_min_jitter_A_pv mos_min_roundtrip_A_pv
A label’s max mos_max_A_pv mos_max_at_A_pv mos_max_packetloss_A_pv mos_max_jitter_A_pv mos_max_roundtrip_A_pv
A label’s average mos_average_A_pv mos_average_packetloss_A_pv mos_average_jitter_A_pv mos_average_roundtrip_A_pv mos_average_samples_A_pv
B labels’s min mos_B_label_pv mos_min_B_pv mos_min_at_B_pv mos_min_packetloss_B_pv mos_min_jitter_B_pv mos_min_roundtrip_B_pv
B label’s max mos_max_B_pv mos_max_at_B_pv mos_max_packetloss_B_pv mos_max_jitter_B_pv mos_max_roundtrip_B_pv
B label’s average mos_average_B_pv mos_average_packetloss_B_pv mos_average_jitter_B_pv mos_average_roundtrip_B_pv mos_average_samples_B_pv
Setting MOS collection on kamailio
set the kamailio config rtpengine params for names the variable the hold specific mos values
Gather the mos stats from the code . Given exmaple is in Lua. The values are filled in after invoking“rtpengine_delete”, “rtpengine_query”, or “rtpengine_manage” if the command resulted in a deletion of the call (or call branch).
KSR.log("info", " mos avg " .. KSR.pv.get("$avp(mos_average)"))
KSR.log("info", " mos max " .. KSR.pv.get("$avp(mos_max)"))
KSR.log("info", " mos min " .. KSR.pv.get("$avp(mos_min)"))
KSR.log("info", "mos_average_packetloss_pv" .. KSR.pv.get("$avp(mos_average_packetloss)"))
KSR.log("info", "mos_average_jitter_pv" .. KSR.pv.get("$avp(mos_average_jitter)"))
KSR.log("info", "mos_average_roundtrip_pv" .. KSR.pv.get("$avp(mos_average_roundtrip)"))
KSR.log("info", "mos_average_samples_pv" .. KSR.pv.get("$avp(mos_average_samples)"))
KSR.log("info", "mos_min_pv" .. KSR.pv.get("$avp(mos_min)"))
KSR.log("info", "mos_min_at_pv" .. KSR.pv.get("$avp(mos_min_at)"))
KSR.log("info", "mos_min_packetloss_pv" .. KSR.pv.get("$avp(mos_min_packetloss)"))
KSR.log("info", "mos_min_jitter_pv" .. KSR.pv.get("$avp(mos_min_jitter)"))
KSR.log("info", "mos_min_roundtrip_pv" .. KSR.pv.get("$avp(mos_min_roundtrip)"))
KSR.log("info", "mos_max_pv" .. KSR.pv.get("$avp(mos_max)"))
KSR.log("info", "mos_max_at_pv" .. KSR.pv.get("$avp(mos_max_at)"))
KSR.log("info", "mos_max_packetloss_pv" .. KSR.pv.get("$avp(mos_max_packetloss)"))
KSR.log("info", "mos_max_jitter_pv" .. KSR.pv.get("$avp(mos_max_jitter)"))
KSR.log("info", "mos_max_roundtrip_pv" .. KSR.pv.get("$avp(mos_max_roundtrip)"))
local mos_A_label = KSR.pv.get("$avp(mos_A_label)")
if not (mos_A_label == nil) then
KSR.log("info", "mos_average_packetloss_A_pv" .. KSR.pv.get("$avp(mos_average_packetloss_A)"))
KSR.log("info", "mos_average_jitter_A_pv" .. KSR.pv.get("$avp(mos_average_jitter_A)"))
KSR.log("info", "mos_average_roundtrip_A_pv" .. KSR.pv.get("$avp(mos_average_roundtrip_A)"))
KSR.log("info", "mos_average_A_pv" .. KSR.pv.get("$avp(mos_average_A)"))
end
local mos_B_label = KSR.pv.get("$avp(mos_B_label)")
if not (mos_B_label == nil) then
KSR.log("info", "mos_average_packetloss_B_pv" .. KSR.pv.get("$avp(mos_average_packetloss_B)"))
KSR.log("info", "mos_average_jitter_B_pv" .. KSR.pv.get("$avp(mos_average_jitter_B)"))
KSR.log("info", "mos_average_roundtrip_B_pv" .. KSR.pv.get("$avp(mos_average_roundtrip_B)"))
KSR.log("info", "mos_average_B_pv" .. KSR.pv.get("$avp(mos_average_B)"))
end
Sample obtained result for avg
INFO: [core/kemi.c:144]: sr_kemi_core_log(): mos avg 3.8 2(260)
SIPp is an opensource (GNU GPL license) performance testing tool for the SIP protocol and is widely used for Quality assurabce of callflows in voip applications for UAC / UASs cenarios. It can emulate functioing of a sip phone such as REGISTER , establishes and releases multiple calls with
the INVITE and BYE methods , send other SIP requests and wait for reponsesbased on dafult of custom xml scenario files .
Plus factor is the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics),
periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management,
regular expressions and variables in scenario files, and dynamically adjustable call rates.
It is widley used as aperformnace and load testing tool since it can test SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, and SIP PBXes and can also emulate thousands of user agents calling your SIP system.
Installation
Pre-requisites to compile SIPp are:
– C++ Compiler
– curses or ncurses library
– For TLS support: OpenSSL >= 0.9.8
– For pcap play support: libpcap and libnet
– For SCTP support: lksctp-tools
– For distributed pauses: Gnu Scientific Libraries
Download , extract and build with options like
tar -xvzf sipp-xxx.tar.gz
cd sipp
./configure --with-sctp --with-pcap --with-openssl
make
Verify installation Run sipp with embedded server (uas) scenario:
sipp -sn uas
On the same host, run sipp with embedded client (uac) scenario:
sipp -sn uac 127.0.0.1 -trace_msg -trace_err
output for server
# sipp -sn uas
------------------------------ Scenario Screen -------- [1-9]: Change Screen --
Port Total-time Total-calls Transport
5060 32.95 s 61 UDP
0 new calls during 0.874 s period 1 ms scheduler resolution
19 calls Peak was 41 calls, after 28 s
0 Running, 63 Paused, 12 Woken up
0 dead call msg (discarded)
3 open sockets
———————————————– 2019-02-04 13:08:13.940159
UDP message sent (371 bytes):
ACK sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-5
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 1 ACK
Contact: sip:sipp@192.x.x.x:5061
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
~ RTP
———————————————– 2019-02-04 13:08:13.941658
UDP message sent (371 bytes):
BYE sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-7
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 2 BYE
Contact: sip:sipp@192.x.x.x:5061
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
———————————————– 2019-02-04 13:08:13.952888
UDP message received [313] bytes :
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-7
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 2 BYE
Contact:
Content-Length: 0
Time
---------------------------- Repartition Screen ------- [1-9]: Change Screen --
Average Response Time Repartition 1
0 ms <= n < 10 ms : 293
10 ms <= n < 20 ms : 9
20 ms <= n < 30 ms : 0
30 ms <= n < 40 ms : 0
40 ms <= n < 50 ms : 0
50 ms <= n < 100 ms : 0
100 ms <= n < 150 ms : 0
150 ms <= n < 200 ms : 0
n >= 200 ms : 0
Average Call Length Repartition
0 ms <= n < 10 ms : 0
10 ms <= n < 50 ms : 0
50 ms <= n < 100 ms : 0
100 ms <= n < 500 ms : 0
500 ms <= n < 1000 ms : 0
1000 ms <= n < 5000 ms : 262
5000 ms <= n < 10000 ms : 0
n >= 10000 ms : 0
------------------------------ Sipp Server Mode -------------------------------
3 new calls during 0.286 s period 1 ms scheduler resolution
0 calls (limit 30) Peak was 25 calls, after 10 s
0 Running, 101 Paused, 7 Woken up
0 dead call msg (discarded) 0 out-of-call msg (discarded)
3 open sockets
----------------------------- Statistics Screen ------- [1-9]: Change Screen --
Start Time | 2019-02-04 13:08:03.908208 1549265883.908208
Last Reset Time | 2019-02-04 13:08:20.954289 1549265900.954289
Current Time | 2019-02-04 13:08:21.241152 1549265901.241152
-------------------------+---------------------------+--------------------------
Counter Name | Periodic value | Cumulative value
-------------------------+---------------------------+--------------------------
Elapsed Time | 00:00:00:286000 | 00:00:17:332000
Call Rate
Tracings
———————————————– 2019-02-04 13:08:13.934840
UDP message received [527] bytes :
INVITE sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-0
From: sipp ;tag=52422SIPpTag001
To: service
Call-ID: 1-52422@192.x.x.x
CSeq: 1 INVITE
Contact: sip:sipp@192.x.x.x:5061
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 135
———————————————– 2019-02-04 13:08:13.948679
UDP message received [371] bytes :
ACK sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-5
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 1 ACK
Contact: sip:sipp@192.x.x.x:5061
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
~ RTP
———————————————– 2019-02-04 13:08:13.949168
UDP message received [371] bytes :
BYE sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-7
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 2 BYE
Contact: sip:sipp@192.x.x.x:5061
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
———————————————– 2019-02-04 13:08:13.949245
UDP message sent (313 bytes):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.x.x.x:5061;branch=z9hG4bK-52422-1-7
From: sipp ;tag=52422SIPpTag001
To: service ;tag=52416SIPpTag011
Call-ID: 1-52422@192.x.x.x
CSeq: 2 BYE
Contact:
Content-Length: 0
time
---------------------------- Repartition Screen ------- [1-9]: Change Screen --
Average Response Time Repartition 1
0 ms <= n < 10 ms : 657
10 ms <= n < 20 ms : 20
20 ms <= n < 30 ms : 0
30 ms <= n < 40 ms : 0
40 ms <= n < 50 ms : 0
50 ms <= n < 100 ms : 0
100 ms <= n < 150 ms : 0
150 ms <= n < 200 ms : 0
n >= 200 ms : 0
Average Call Length Repartition
0 ms <= n < 10 ms : 649
10 ms <= n < 50 ms : 28
50 ms <= n < 100 ms : 0
100 ms <= n < 500 ms : 0
500 ms <= n < 1000 ms : 0
1000 ms <= n < 5000 ms : 0
5000 ms <= n < 10000 ms : 0
n >= 10000 ms : 0
------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause traffic -----
Last Error: Overload warning: the major watchdog timer 3000ms has been t…
Controlling sipp
[+|-|*|/]: Adjust rate
p Pause traffic
q Quit SIPp (after all calls complete, enter a second time to quit immediately)
Q Quit SIPp immediately s Dump screens to the log file (if -trace_screen is passed)
GStreamer ( LGPL )ia a media handling library written in C for applicatioan such as streaming , recording, playback , mixing and editing attributes etc. Even enhnaced applicaiosn such as tsrancoding , media ormat conversion , streaming servers for embeeded devices ( read more about Gstreamer in RPi in my srticle here).
It encompases various codecs, filters and is modular with plugins developement to enhance its capabilities. Media Streaming application developers use it as part of their framework at either the broadcaster’s end or as media player.
gstreamer-bad-video-1.0GStreamer bad video library – Bad video library for GStreamer elementsgstreamer-tag-1.0 GStreamer Tag Library – Tag base classes and helper functionsgstreamer-bad-base-1.0 GStreamer bad base classes – Bad base classes for GStreamer elements
gstreamer-audio-1.0 GStreamer Audio library – Audio helper functions and base classes
gstreamer-plugins-bad-1.0 GStreamer Bad Plugin libraries – Streaming media framework, bad plugins libraries
gstreamer-rtsp-server-1.0 gst-rtsp-server – GStreamer based RTSP server
At the time of writing this article Gstreamer an much early version in 1.X , which was newer than its then stable version 0.x. Since then the library has updated many fold. summarising release highlights for major versions as the blog was updated over time .
Gstreamer 1.8.0 – 24 March 2016
Features Hardware-accelerated zero-copy video decoding on Android
New video capture source for Android using the android.hardware.Camera API
Windows Media reverse playback support (ASF/WMV/WMA)
tracing system provides support for more sophisticated debugging tools
high-level GstPlayer playback convenience API
Initial support for the new Vulkan API
Improved Opus audio codec support: Support for more than two channels; MPEG-TS demuxer/muxer can handle Opus; sample-accurate encoding/decoding/transmuxing with Ogg, Matroska, ISOBMFF (Quicktime/MP4), and MPEG-TS as container; new codec utility functions for Opus header and caps handling in pbutils library. The Opus encoder/decoder elements were also moved to gst-plugins-base (from -bad), and the opus RTP depayloader/payloader to -good.
Asset proxy support in the GStreamer Editing Services
GStreamer 1.16.0 – 19 April 2019.
GStreamer WebRTC stack gained support for data channels for peer-to-peer communication based on SCTP, BUNDLE support, as well as support for multiple TURN servers.
AV1 video codec support for Matroska and QuickTime/MP4 containers and more configuration options and supported input formats for the AOMedia AV1 encoder
Closed Captions and other Ancillary Data in video
planar (non-interleaved) raw audio
GstVideoAggregator, compositor and OpenGL mixer elements are now in -base
New alternate fields interlace mode where each buffer carries a single field
WebM and Matroska ContentEncryption support in the Matroska demuxer
new WebKit WPE-based web browser source element
Video4Linux: HEVC encoding and decoding, JPEG encoding, and improved dmabuf import/export
Hardware-accelerated Nvidia video decoder gained support for VP8/VP9 decoding, whilst the encoder gained support for H.265/HEVC encoding.
Many improvements to the Intel Media SDK based hardware-accelerated video decoder and encoder plugin (msdk): dmabuf import/export for zero-copy integration with other components; VP9 decoding; 10-bit HEVC encoding; video post-processing (vpp) support including deinterlacing; and the video decoder now handles dynamic resolution changes.
ASS/SSA subtitle overlay renderer can now handle multiple subtitles that overlap in time and will show them on screen simultaneously
Meson build feature-complete (with the exception of plugin docs) and it is now the recommended build system on all platforms. The Autotools build is scheduled to be removed in the next cycle.
GStreamer Rust bindings and Rust plugins module
GStreamer Editing Services allows directly playing back serialized edit list with playbin or (uri)decodebin
Market trends are really not in favor of Telecom Service /providers with increasing use of OTT ( Over The Top ) application like watsapp , Facebook messenger , Google hangouts , skype , viber , etc .
OTT ( Over The Top ) Applications
What is an OTT ?
An Over The Top ( OTT ) application is one which provides communication services over Internet . Therefore these bypass the communication billing system setup by a Telecom Operator , resulting in no gain or loss of revenue to Telecom Operator who is providing the Internet service to user in first place .
Hence we see that OTT are major threat and concern for Telecom Operators whose traditional and obviously expensive ( when compared to OTTs free service ) billing models are facing disruption .
Telecom Regulatory bodies around the world
The telecom regulatory authorities in some of the countries are for example listed as :
Canadian Radio-television and Telecommunications Commission (CRTC) – Canada
Ministry of Information Industry (MII) – China
Autorité de Régulation des Communications Électroniques et des Postes (ARCEP) – France
Bundesnetzagentur (BNA) – Germany
Telecom Regulatory Authority of India (TRAI) – India
Ministry for Communications and Informatization of the Russian Federation (Minsvyaz) – Russia
Infocomm Development Authority of Singapore (IDA) – Singapore
Independent Communications Authority of South Africa (ICASA) – south Africa
Federal Communications Commission (FCC) , National Association of Regulatory Utility Commissioners (regulators of individual states) (NARUC) , CTIA – The Wireless Association (CTIA) – USA
Such telecom regulatory bodies get to decide whether to enforce differential price to end consumers for using OTT so that telecom service providers can benefit or keep the Internet fair and open by passing Net Neutrality Laws and Bills and amendments .
what is Net Neaurality ?
The fundamental principle of Net Neurality is that Telecom Operators should not block , slow down or charge consumers extra for using other services as their means of communication. This states that it is wrong to charge users above the regular data rates for using VOIP apps and other internet based communication services.
The following counteries have adopted principles of Net Neutrality by passing bills or making law .
Chile – Chile’s General Law of Telecommunications, “No [ISP] can block, interfere with, discriminate, hinder, nor restrict the right of any Internet user of using, send, receive, or offer any content, application, or legitimate service through the Internet, as well as any activity or legitimate use conducted through the Internet.”
Brazil – ” Internet Bill of Rights ” makes equal access to internet mandatory in Brazil .
Netherlands – Even European Union has adopted Netherlands’ Net Neutrality amendment which reads “traffic should be treated equally, without discrimination, restriction or interference, independent of the sender, receiver, type, content, device, service or application.”
USA – Citizens make ‘We the People’ platform to ‘Restore Net Neutrality By Directing the Federal Communications Commission (FCC) to Classify Internet Providers as ‘Common Carriers‘. Therefore not allowing them to either throttle speed by paid prioritization , discriminate in pricing or block any broadband access to legal content . Above facts are from this tech.firstpost.com article.
Inspite of the fact that I Support Net Neutrality with all my heart , as a telecom engineer I understand the cost investment made by Telecom operators in providing am efficient communication network to its subscribers ( Access , Network and Application layers ). Therefor I do have my sympathies with the Telcos and to level out the wide ranging conflict between Telcos and ISP ( Internet Service Providers ) , I pen down the following points which reflect the Telecom Operators Problems and also highlight the solutions that can be adopted to counteract the OTT threat .
Depleting revenue for Telco
Messaging – OTT messaging cost operators $13.9 billion, or 9% of message revenue in 2013
Voice – Voice services under threat from VOIP services like Skype, Viber
OTT apps – Voice & Message apps have been the operator’s biggest headache. Its time Operator should launch its own OTT Services
Data Traffic – The utilization is yet to reach its peak. Will face challenges from WiFi access
Critical Pain areas – Erosion of Operator’s revenue from voice and (especially) messaging
Telco’s OTT aPPLICATION
At this stage it is crucial for a telecom Service provider / Operator to enter the Apps market and bring forth a Messenger which is more powerful , interactive and awesome than a OTT application. Fortunately the Operator can always couple this application with his background telecom infrastructure to provide the edge in performance and functionalists .
Road block while developing a OTT application for a Telecom Service Provider :
Investment in Data Network is not being utilized due to lack of service
Reuse of Existing business Logic and extending the service reach across devices and networks is tough
Operator already has full fledged network Infrastructure in Place
Desire for minimum CAPEX while investing in new technologies
compete with OTT players and open new revenue streams is a challenge
Next we find the way of solving the problems and integrating them together to form a Solution .
OTT Application for Telecom Service provider
Introduce new services to benefit from investment on Data Plans and Bandwidth
Expose REST API to enable 3trd party Integration with existing network Infrastructure
Partner with individual OTT players to make new services that do not compete on core competencies like billing etc
Use protocols like SIP that reduce CAPEX and have goto market more quickly
Go for enriched service that lead to better user experience
This writeup outlines the process of creating a OTT application for a Telecom Service Provider . Components for the application include cloud Address Book , Video Chatting , Location share , Contact synchronization ,REST based thin client , OS and device agnostic etc shown in the figure below
telco’s OTT app
The Application is designed to close knit with Operator’s own infrastructure hence the crucial entities like Network Address Book , Location Service are synced and fetched from Backend Network .
OTT application Feature Overview
Smart Address Book
Automatic: Get contacts from Gmail, Facebook
Fast search by first, last name, frequently
dialed number
Roadmap: View calendar events
Personal: Get image from Gmail and display in contacts list
Geo Location
Share own location during chatting
Get map for calculating the distance between two chat users
Roadmap : Trigger device (say Switch on/off AC before reaching home) from a threshold distance away from home location
Messaging
Ad-hoc Chat
Session Based Chat
Voice Input for texting
Presence information of contacts
RoadMap: Legacy message integration
Telephony
Voice call to mobile
Voice call to PSTN
Video call to other @imAll user
Share images during voice call to other
Device agnostic
Compatible with IOS, windows
Can run as native app on ipad
Can run as browser client on windows
RoadMap: native app for android, windows phone,blackberry10
Roadmap
To upgrade the application and provide enganced and enrich service support the I propose the following roadmap.
From plain vanilla voice and video calling ( supported by every other OTT application ) our application should progress towards legacy telecom support whihc included PSTN , GSM , ISDN etc . This requires backbone of telecom network and a good setup for media codec conversion to suit various legacy media codecs .
Road Map from Traditional to New age services
Voice and video calling
Legacy services support like MMS and SMS
Integration with 3rd party Vendors
Give new enriched services like Multilingual support , file transfer , screen-sharing etc
give facility to integrated web plugins for web calling
To keep the interest of customers it is essential that the application be supported on other popular OTT services like skype , Gtalk . for exmaple a caller should be able to make call from Skype / Gtalk to our application .Multilingual capabilities, support for larger protocol spectrum will just act like icing on the cake .
How does it benefit the Operator??
Saves on development cost and time
Device Agnostic OTT Applications
Simplified Service deployment
Saves licensing cost per client
Reuses existing Messaging and Address Book service logic.
Open New Revenue Streams for operator
No separate SIP stack required for the client
Faster Time to Market
Update : At the time of writing this post I did not anticipate the wave of change that bring focus on subjects like “net neutrality” , ” Save the internet” and “free internet” . However I see now that I had described this phenomenon way in advance for my time .
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.
I have been contemplating points that make for a successful developer to develop solutions and services for a Telecom Application Server. The trend has shown many variations from pure IN programs like VPN , Prepaid billing logic to SIP servlets for call parking , call completion. From SIP servlets to JAISNLEE open standard based communication.
I can assert with confidence that Telephony is eventually going to be integrated with Internet to provide a new breed of communication in future aka Internet Telephony.
I have assembled some points here that make for a successful startup in logic building domain of telecom core network .
1. Product Innovation
Envisions a multiple network technologies, that provides ability to build over new innovative cutting edge technologies in the market. It should deliver platform to launch newer services like WebRTC and RCS .
2. Competitive Pricing
Cutting down the spiraling cost of Development of the new technologies platform with improvement in the usage of Data rather than voice by integrating new features like File sharing and MSRP messaging. An evolutionary architecture to reduce the effort and cost through high re-use of NGN Platform and Services.
3. Lower Servicing costs
Introduce uniform service experience across different platforms which helps CSP’s to reduce Time Cycles and Costs for handling enhancements requests and the annual OPEX appreciably.
4.Flexibility
Interworking among the services from legacy IN solution and IMS /IT. Allow the Operators to extend their basic offering with added services via low cost software and increases the ARPU for subscribers.
5. Develop on webRTC
Make a WebRTC solution that offers a plug in free, device agnostic, network agnostic web based communication tool along with the server side implementation.
6. Media solutions
CSP’s are looking into Long term growth and profitability from new online services media streaming services . Make use-cases around IPTV and VOD ( Video On Demand) . Also Voicemails , IVR , DTMF, TTS( text to speech ) , Speech recognition etc
7. Social Media Integration
Integration of the services with social media/networking enables new monetizing benefits to CSPs especially in terms on advertising and gaining popularity , inviting new customers etc.
Enterprises are looking forward to reach customers with ennoblement of Telco in their present landscape which was impossible to reach before. Telco not only plays an instrumental role in increasing the customers base which results into increase in enterprise’s revenue but also offers the value addition in their present product/service delivery model. Hence it is high-time when developers can aggregate , use open-standard services / technologies ( GSMA , SIP , WebRTC ) and develop high end solutions for Telecom Domain .
