A second application of voiceoverpacket, shown in Figure 2, is a trunking application. In this scenario, an organization wants to send voice traffic between two locations over the packet network and replace the tie trunks used to connect the PBXs at the locations. This application usually requires the IWF to support a higher capacity digital channel than the branch application, such as a T1/E1 interface of 1.544 or 2.048 Mbps. The IWF emulates the signaling functions of a PBX, resulting in significant savings in companies' communications costs.
An off-net call incoming from a non-CGVoP network is first detected at the Signaling Gateway. Call setup information (a type of communication-identifying information) passes from the Signaling Gateway to the CMS. The CMS uses the NANP DN number it receives to identify its internal network address for that DN, determines where the call should terminate (commonly from registered address information) and the characteristics of the connection, and initiates ringing to alert the end-user of the incoming call. When the voice content packets arrive at a CGVoP network’s Trunking Gateway from the PSTN, the packets are converted from circuit mode to packet mode. Conversion of the voice-band transmission to packet transmission and connection to the end user’s CPE are the functions of the gateways and routers in the CGVoP network. As with outgoing calls, the CMS typically involves itself in an active call only when a recognized feature is invoked by the end user.
connectionless network, the individual packets of each voice signal may travel over separate network paths for reassembly in the proper sequence at their ultimate destination. While this provides for a more efficient use of network resources than the circuit- switched PSTN, which routes a call over a single path, it also increases the chances for packet loss. Since all voice frames are treated as data, packets may be dropped under peak loads and during periods of congestion (caused by link failures or congestion). Due to the time sensitivity of voice transmissions, the normal Transmission Control Protocol (TCP) based retransmission schemes are not suitable. Packet losses greater than 10% are generally not tolerable. The data frames, however, are not time-sensitive and dropped or erroneous packets can be appropriately corrected through the process of retransmission. Lost voice packets, however, cannot be dealt with in this manner. Some schemes used by voiceoverpacket software to address the problem of lost frames are given in  as: • Interpolate for lost speech packets by replaying the last valid packet received during
distortion (THD) among others. These and other linear measurements are useful only in certain cases because they assume that changes to the voice waveform represent unwanted signal distortion. These testing methods also assume that telephony circuits are essentially linear. However, in VoIP and other voice-over-packet networks, particularly when low bit rate speech-codecs such as G.729 and G.723.1 are used, neither waveform preservation nor circuit linearity can be assumed. These codecs try to reproduce the subjective sound of the signal rather than the shape of the speech waveform, rendering traditional testing methods more or less ineffective. And, as discussed before, the bursty and time-insensitive nature of packet networks exposes the need for other testing methods as well. Finally, because of their heightened importance, the performance of echo cancellers, voice activity detectors, and other processes needs to be tested directly.
Voiceover IP or telephony via the Internet, characterizes a set of protocols/technologies [H.323, Session Initiation Protocol (SIP)], which provide vocal conversation in real time with relatively good quality at little or no cost, thanks to worldwide broadband connections. Traditionally, such conversations took place exclusively through a PC which was connected to the Internet and with the aid of a microphone, headphones and the appropriate software (soft phones). The call ended up in another similarly equipped PC at no extra charge other than the one needed to access the Internet, since this specific type of communication does not require a provider of standard land-line services, but only the Internet. In addition to this, there are autonomous telephone devices (VoIP SIP Phones) and analog telephone adaptors (ATA) on the market which directly connect to an IP network, like the Internet. With the right adjustments and assembly and without the need of a PC, use of this service is facilitated making VoIP even more accessible to its users.
functioning of modems by confusing the constellation encoding used. The result could be modems that never synchronize or modems that exhibit very poor throughput. Some gateways might implement some intelligence that can detect modem usage and disable compression. Another potential issue deals with low-bit-rate speech compression schemes, such as G.729 and G.723.1. These encoding schemes try to reproduce the subjective sound of the signal rather than the shape of the waveform. A greater amount of packet loss or severe jitter is more noticeable than that of a non-compressed waveform. However, some standards might employ interleaving and other techniques that can minimize the effects of packet loss.
Richard Olsen, B.S.E.E., P.E. Richard Olsen holds a B.S. in Electrical Engineering and has over 36 years of professional engineering and teaching experience. Richard held senior management and engineering positions at Southwestern Bell for over 20 years. He has served as an instructor for numerous major companies as well as being an Adjunct Professor and Executive-in-Residence in the Master of Science in Telecommunications Management degree program, Oklahoma State University. Richard is a member of the MSTM Industry Advisory Board, Oklahoma State University, the National Society of Professional Engineers, the Oklahoma Society of Professional Engineers and the IEEE. Richard consistently receives excellent ratings across the board on student evaluations, with many comments specifically praising his knowledge and thoroughness in making sure everyone understands the topics.
There is no frequent background ‘keep-alive’ traffic associated with IMS based services, which is an advantage over OTT VoIP services. OTT VoIP apps (Viber, Skype, etc.) must maintain active sessions with keep-alive messages in order to stay reachable for incoming calls. These frequent keep-alive transactions eventually result in large signaling load in the network. With IMS services, the client device performs periodical re-registrations to the IMS, but the frequency is significantly lower than that of OTT apps. A VoLTE service requires additional signaling for setting up a dedicated GBR bearer to fulfill the QoS requirement, but the expected network impact is low. It is worth noting that smartphone platforms such as iOS, Android and Windows Phone tend to have platform specific connections (e.g. to get notifications and automatic software updates), which generate data transactions and signaling load. The impact of VoLTE on signaling load is therefore assumed to be negligible.
In many deployments the IMS/VoIP capable radio coverage may be initially less extensive than the concurrent Circuit Switched (CS) voice coverage. In order to offer its VoIP customers a seamless voice service already at that stage, the operator may wish to utilize the CS radio access as a complement to the IMS VoIP capable radio coverage. This Annex describes the features for the UEs and networks that wish to support such a deployment scenario, need to implement, in addition to the IMS VoIP over LTE minimum feature set.
For over 100 years, telephones have grown to become a primary means of communications in both our personal and business lives. Even with all the changes from analog to digital, wireline to mobile and eventually to Voiceover IP, one thing has remained consistent – limited audio quality. Why do we have to sound like air traffic controllers when spelling out confirmation codes? “Papa, Alpha, Delta”. This is due to the limitations that the PSTN enforces on traditional analog and digital telephones and the “3.4 kHz sound barrier”. The adoption of VoIP and broadband networks have given us the opportunity to break through this barrier with a whole new range of wideband and high-quality voice coding algorithms that make communications more efficient, effective and natural. HD VoIP allows carriers to differentiate their services with a much improved audio experience, creating customer loyalty and affinity. Enterprises can differentiate themselves with superior voice quality to their customers, building on their quality branding while improving business efficiency. AudioCodes HD VoIP solutions and products - the way sound was meant to be heard.
Thus, assuming a range of 4000 Hz, this requires a rate of 8000 samples per second. Remember that each sample is assigned an 8-bit value to represent the amplitude height at the time of sampling. Thus, a dedicated 64,000-bit channel (8-bits x 8000 samples per second) was traditionally required for a voice call (hence a DS0 being 64Kbps).
Step 4. The Intel Dialogic Configu- ration Manager should appear show- ing one board detected. At this point the drivers for the board have not been started so the board icon appears with a red square. To start the voice board drivers, click the green arrow button as shown.
ABSTRACT:Voice over IP (VOIP) is an upcoming technology that enables voice communication through the Internet. Packet-based network link shared between different connections, which give rise to interaction between various traffic types. This paper focus is on improving the data embedding capacity of steganography in low bit rate audio streams encoded by G711 source codec and to overcome the packet loss during the integration of hidden messages. Steganography in the inactive audio frames attains a larger data embedding capacity than that in the active audio frames under the same imperceptibility. The amount of data package increases when new services are offered and used via the internet. The large amount of data packages, the more information let through and new possibilities to hide information i.e., in the cover of something else, may be introduced. To identify the voice during the transmission whether the current audio frame is an active voice by comparing the energy of the frame with a threshold.
C. Second Simulation Parameters and Network Topology Three scenarios are implemented; three areas are operated at each scenario area three for the DRX disabled mode, area (2) for the DRX mechanism short cycle mode and area (1) for the DRX mechanism long cycle mode. Three mobile application users are used; the first user rotates around the area (3) where DRX disabled in the first scenario. On the other hand, the second user rotates around the area (2) where Short cycle DRX (10 msec) in the second scenario and the third user rotates around the area (1) where Long cycle (40 msec) DRX in the third scenario at the same time. The simulation run time =490 seconds, in the three scenarios the three modes differentiated. The network consists of three eNodeBs, parameterized, according to Table I, II, III, and IV. Concerning mobility, three Mobile users, and structure also includes elements EPC (Evolved Packet Core) and gateway that will communicate with the Four-application server, as shown in figure 26. Three types of data used (Voice, Video, and HTTP), performance matrices such as (End to End delay, Packet delay variation, MOS, and Traffic sent received) used in Voice and Video traffic, Object and Page response time for HTTP Traffic, SNR, BER, Throughput, PUSCH, PDSCH, Downlink and Uplink Packet drop for LTE-A network. Fig.22 UL Packet drop IDLE and DRX mechanisms
Most of the packet losses come from the transmission failure of the AP. The reason is that 802.11b is designed in a way that every node has to wait for a random amount of time before it tries to send a packet. When the data is concentrated into the AP in the center, at some instant the AP will hold many packets that need to be injected in the network. However, there still have other nodes that are trying to flood packets, so in a fair manner, these packets will be accumulated in queue. If the state of the queue is defined as the number of waiting packets in the queue, this queuing system is unstable. It will eventually overflow and start to drop packets by a Drop Tail manner, which is the default setting in the simulation.
may vary. The receiver buffers the early-arriving packets and waits for the latecomers to catch up, which typically takes 20-40 milliseconds (or more). Many of the above factors, such as WAN propagation delays, are fixed and cannot be improved. Others, such as the choice of signal encoder, are options that can affect network performance. Many encoding algorithms have been designed, some of which are published as ITU-T standards, and others that are proprietary to a particular vendor. These techniques differ in their underlying mathematical algorithms plus their end results. In general, algorithms that compress the voice (and thus conserve bandwidth) have higher processing delays and lower Mean Opinion Scores (MOS) than algorithms that use no compression. For example, the ITU-T G.711 standard uses a Pulse Code Modulation (PCM) technique and transmits at 64 Kbps with a negligible coding delay. The G.723.1 standard uses two techniques: Algebraic Code Excited Linear Prediction (ACELP) at 6.3 Kbps, or Multipulse Maximum Likelihood Quantization (MP-MLQ) when operating at 5.3 Kbps, with a coding delay of 37.5 milliseconds. A third standard, G.729, uses Conjugate Structure Algebraic Code Excited Linear Prediction (CS-ACELP) coding and operates at 8 Kbps with a coding delay of 15 milliseconds, providing a middle point between these two extremes.