(PMICT) payam e mehr Iran

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General office

(PMICT) payam e mehr
no 75 - kheradmand st - karimkhan st - 7 tir sq
tehran
tehran
Iran

Phone: 00982188863655
Phone: +982188863661
Site: http://pmict.ir

7, 10 and 11 digit dialling
A-Z Termination
Agent Dashboard License
Agent Virtualization
Alternate Caller ID
Billing and Routing Solution
call centers
Cloud-based Unified Communications Service Provider
did numbers provider
FREE Callshop Software
Hardware
Hosted PBX Service Provider
Hosted VoIP billing service provider
Incumbent PTT
International VoIP Wholesale Provider
Internet Fax Service Provider
Internet Telephony Service Provider
Internet VoIP and Video Conferencing Service Provider
Marketing Services
Mobile VoIP
SIP Billing
VoIP Billing Software Provider
Voip consulting
Voip Termination ISP
Wholesale VoIP Carrier

Alternate Caller ID
Alternate Routing Options
Analog Adapters
Anonymous Call Rejection
Answering Rules
Audio Codec’s: G.729, G.711
Auto-attendant
Automatic Call-back
Billing Software
Blacklist
Busy Lamp Field (BLF)
BYOD support
Call Back
Call Center services
Call Controller
Call filter
Call Forwarding
Call Handling
Call Intercept
Call Logs
Call Me Now
Call Relay VoIP Solutions
Call Routing VoIP Solutions
Call shop solution
Call Transfer
Call Waiting
Call Waiting With Caller ID
Caller ID
Caller ID Blocking
Caller ID Control
Classes of Service
Click To Call
Click to Call Me
Cloud PBX
Codec
Collaboration
Conference Bridge
Conference Phones
Conferencing
Consultancy Services
CRM Migration
Custom Greetings
Custom Ring Back
Customizable Greetings
Data Conferencing
Desk Phones
Detail & Phone Usage Summary Reports
Dial-by-Name Directory
DID Support
Directed Call Pick Up
Directory Number Hunting
Distinctive Ring
DNIS Support
Do Not Disturb
E1 Bulk Wholeseller
E1/T1
E911 Support
Email to Fax, and vice versa
Enhanced Outgoing Calling Plan
Enhanced Voicemail
Entrance Greeting and Queue Announcements
Extentions
External/Internal Transfer
Failover
Fax To Fax
Find Me, Follow Me
Flash on Hold
Forced Delivery of Calls
Free In-Network Calling
Gatekeepering
Greetings
Group Paging
gsm voip solutions
H.323 Softswitch & CPE
H.323 VoIP Gateway
H.323 Wireless/ GSM VoIP Solutions
Hosted PBX
Hosted VoIP

3CX Phone System
Antek
Asterisk
Asterisk
Asterisk
Audiocodes
Cisco Systems
Cisco Systems
Digium
DSG
grandstream
Huawei VoIP Products
Ingate
iTele Switch
net.com
Netvox
SIP softphones
voipswitch

a2billing
Billing system of own development
IP PBX
IP Phone
iTel Billing

SAMAND Communication''s

Click Voip

Providers in database: 6433

RTP Delivers Multimedia Data In Real Time

A Real-time Transport Protocol (RTP) is a protocol standard defined by the Internet Engineering Task Force (IETF) that outlines a management system for programs using real-time transmission of data through mutlicast or unicast network services in multimedia situations. Originally designed by the IETF for videoconferencing for multiple participants, the protocol is largely used in Voice over Internet Protocol applications. Despite its name, the protocol cannot guarantee real-time delivery of data, however, the RTP does compensate for jitters and can detect when data arrives out of sequence, both issues being popular in VoIP communication. In IP telephony, RTP works with a signaling protocol, such as SIP or H.323, in order to set up connections in a network.

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