- Characteristics of the Default Dial Peer 177 Matching Outbound Dial Peers 179 Summary 180. - Cisco Unified Communications. - This book is primarily targeted toward candidates of the CVOICE exam. - The header portion consists of the IP segment, the UDP segment, and the RTP segment. - On the other hand, you might prefer H.323 over MGCP because of the wider selection of interfaces supported.. - The integrated services routing architecture of the 3800 Series is based on the 3700 Series. - The Cisco 827-4V ADSL Router is a member of the Cisco 800 Series Routers. - Each IP Telephony deployment model differs in the type of traffic that is carried over the WAN, the location of the call-processing agent, and the size of the deployment. - The main benefits of the single-site model are the following:. - Single-site deployment is a subset of the distributed and centralized call-processing model. - Criticality of the voice network. - For specific sizing recommendations, refer to the Cisco Unified Communications SRND based on Cisco UCM 6.x at the following link:. - Use of the IP WAN to bypass toll charges by routing calls through remote site gate- ways, closer to the PSTN number dialed (that is, tail-end hop-off [TEHO]). - The following best practices apply to the use of SIP proxies:. - The main factors, for the pur- pose of design, are the size of the site and the functionality required.. - Cisco UCM 50 to 30,000 phones Small to large sites, depending on the size of the UCM cluster.. - Clustering over the WAN involves having the applications and UCM of the same cluster distributed across the IP WAN.. - Benefits of the Clustering over the IP WAN Deployment. - The following design guidelines apply to the indicated WAN characteristics:. - Which two of the following signaling protocols are peer-to-peer protocols?. - Which of the following best describes a function of RTCP?. - Which two of the following VoIP gateway platforms are considered to be Integrated Services Routers (ISRs)? (Choose 2.). - Which four of the following are Cisco-supported IP telephony deployment models?. - However, com- pressing voice can degrade the quality of the voice. - The varying arrival time of the packets can cause gaps in the re- creation and playback of the voice signal. - Delay: Delay is the time between the spoken voice and the arrival of the electroni- cally delivered voice at the far end. - I want you,” the listener might hear, “Wat....s...on...come here, I...wa...nt...y...ou.” The variable arrival of the packets at the receiving end causes the speech to be delayed and garbled.. - I want you,” the listener might hear, “Wat. - The test results are subjective, because they are based on the opinions of the listeners. - The measurement is made by comparing the original transmitted speech to the resulting speech at the far end of the transmission channel. - Voice traffic is also intolerant of packet loss and jitter, both of which unacceptably degrade the quality of the voice transmission delivered to the recipient end user. - “priority” bandwidth (if the voice traffic needs that much bandwidth), meaning the voice traffic is transmitted first, ahead of the web traffic. - Fax pass-through is the state of the channel after the fax up-speed process has occurred.. - and turn off echo cancellation and VAD for the duration of the call.. - Data is packetized and encapsulated according to the T.38 standard. - Switching to the G.711 codec. - Redundancy can be enabled in one or both of the gateways. - The following are some of the basic characteristics of on- and off-ramp faxing:. - For the duration of the call, the DSP listens for the 2100-Hz CED tone to detect a fax or modem on the line.. - The call control stack on the OGW instructs the DSP to send an NSE to the TGW, informing the TGW of the request to carry out a codec change.. - The DSP in the Cisco IOS gateway attached to the fax machine that gener- ated the DIS message (normally the TGW) detects the High-Level Data Link Control (HDLC) flag sequence at the start of the DIS message and initiates fax relay. - The encoding of the packet headers and the mechanism to switch from VoIP mode to fax relay mode are clear- ly defined in the specification. - The DSP in the Cisco IOS gateway attached to the fax machine that generated the DIS message (normally the TGW) detects the HDLC flag sequence at the start of the DIS message and initiates fax relay switchover. - If the DSP on the gateway is capable of supporting T.38 mode, this information is indicated during the H.245 negotiation procedures as part of the regular H.323 VoIP call setup.. - When a fax tone is heard, the DSP signals the receipt of the fax tone to the call control layer, which then initiates fax changeover as specified in the T.38 Annex B procedures.. - At the end of the fax transmission, another INVITE message can be sent to return to VoIP mode.. - transparent to the call agent. - A fax relay MGCP event allows the gateway to notify the call agent of the status (start, stop, or failure) of T.38 processing for the connection. - This method requires the use of Cisco gateways at both the originating and terminating endpoints of the H.323 call.. - That transmission is transparent to the call agent. - There are two subsets of the G.711 codec:. - After insertion, the media streams are connected between the MTP and the H.323 device, and these connec- tions are present for the duration of the call. - The media streams connected to the other side of the MTP can be connected and disconnected as needed to implement features such as hold and transfer.. - The DSP on the PVDM2-8 has one-half the capacity of the DSP used on other PVDM2 modules. - Clicking this link provides a template of the Cisco IOS configuration to be applied to the router to support the DSP media resources.. - To reset to the default, use the no form of the command.. - Global configuration includes the configuration of the individual Cisco Unified Communications Managers, the local SCCP interface used for signaling, and activating SCCP.. - Which of the following media resources require DSPs (that is, the resource cannot be performed by Cisco Unified Communications Manager)?. - The applications discussed help illustrate the function of the voice ports, whose configuration is addressed in the next section.. - In many instances, the type of port is dependent on the voice device connected to the network. - IP phones that connect to the network via switches place on-net calls through Cisco Unified Communications Manager. - The call is sent from the local voice-enabled router that is acting as a gateway to the PSTN. - The call is then sent to the PSTN for call termination.. - The call is automatically dialed based on the PLAR configuration of the voice port. - This includes IP phones connected to the IP network. - The output of the WAN gateway is either down or congested, so the call is rerouted.. - The call connects to the PSTN.. - The PSTN completes the call to the remote site.. - The call is sent from the local voice-enabled router, which acts as a gateway, to the PSTN. - The following is a detailed explanation of each of the three types of analog voice interfaces:. - The router acts as this side of the interface. - Loop-start, as shown in Figure 3-10, is the more common of the access signaling tech- niques. - On the voice gateway, the FXO port sends address signaling to the FXS port. - Users should be familiar with most of the following call progress tones:. - There are six distinct physical configurations for the signaling part of the interface. - After a timed interval, the calling side looks at the status of the called side. - An FXO trunk is one of the simplest analog trunks available. - Both sides of the trunk need to have a matching configuration. - Consider the following characteristics of the trunks depicted in Figure 3-18:. - If a subscriber at the London site places a call to the PSTN, the gateway uses one voice channel of the E1 R2 trunk interface.. - Table 3-5 presents a description of the common analog trunk features.. - You can configure this to match the require- ments of the connected device.. - Dial Peer). - In Figure 3-25, the telephony device connects to the Cisco voice-enabled router. - The POTS dial-peer configuration includes the telephone number of the telephony device and the voice port to which it is attached. - The Cisco voice-enabled router VoIP dial peer is connected to the packet network. - In a small network environment, the device might be the IP address of the remote device. - Use the session target command to specify the IP address of the terminating router or gateway.. - The session target ipv command defines the IP address of the router connected to the remote telephony device.. - Create VoIP dial peers for each of the R1 and R2 sites based on the diagram presented in Figure 3-29.. - Note Cisco IOS Software does not check the validity of the E.164 telephone number. - Inbound POTS dial peers are associated with the incoming POTS call legs of the originating router or gateway.. - Inbound VoIP dial peers are associated with the incoming VoIP call legs of the ter- minating router or gateway.. - Port Attempts to match the configured dial peer port to the voice port associated with the incoming call (POTS dial peers only).. - The router or gateway attempts to match the called number of the call setup request with the configured incoming called-number of each dial peer.. - If a match is not found, the router or gateway attempts to match the calling number of the call setup request with the answer-address of each dial peer.. - If a match is not found, the router or gateway attempts to match the calling number of the call setup request to the destination-pattern of each dial peer.. - All support calls are routed to the same trunk group destined for the call center. - Characteristics of the Default Dial Peer. - Dial Peer 2
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