Pay attention that normally the order of the partitionins is always L1,L2,L3, D1,D2,D3 , where Lx are the partitions assosiated to the Line model and Dx are the partitions associated to the Device model.
So this order will give you the order/ priority in case of tie-break !
In the case of CTI Route Point or CTI ports , the order is changed by the following D1,D2,D3,L1,L2,L3 …. Please recall this if you are involved in a troubleshooting with these elements !
Here is the list with all the Call Manager services:
- Call Manager : Provides call processing
- TFTP : responsible for building and serving files for devices
- Messaging Interface : used with voice mail systems which use SMDI
- IP Voice Media Streaming Application: provides functionality underlying MOH, conferences, MTP, …
- Telephony Call Dispatcher (TCD) : provides hunting group with Auto-Attendant console
- CTL Provider : works with CTL Client to change security mode from nonsecure to secure
- MOH Audio Translator : it converts audio files for MOH utilisation
- RIS Data Collector : collects and distributes real-time information
- Extension Mobility : login/logout features
- CDR Insert : is responsible for CDR operations
- IPMA : Manager/assistant service
- Extended functions : call-back and QRT service
- Serviceability Reporter : generates reports on logged information
- Webdialer : allows web application to make call control
- CAPF : Issues locally Significant Certificates (LSC)
TFTP is the part of Call Manager where you store all phone’s configuration so that you are able to retrieve then at the bootup sequence. TFTP IP Address will be communicated to IP Phones via the DHCP option 150.
There is two types of security that you can enable with Call Manager
- Mixed mode : In this mode, depending the security configured on each phone, you can have secure calls when both devices are security-enabled and when one of the phones is missing security, your call will be nonsecure.
- Nonsecure mode : As all phones are not set up with security (default configuration), all calls are nonsecure.
When you device to put security on phones , they can support the three following levels:
- Nonsecure mode : secure calls are not supported
- Authenticated mode : the phone will be able to authenticate calls
- Encrypted mode : the phone will be able to support encrypted calls
If you enable the authentication and the encrytion on your network , you are then able to secure the media traffic as well the voice signaling.
If you want to have security on the media flow, it is then mandatory to secure also the signaling as the keys which are used to secure the media traffic are exchanged during the signaling phase.
SCCP messages sent by IP Phones and Call Manager can be secured using TLS, it is the signaling part. Then for the protection of the media traffic so the RTP packets , you will use the Secure RTP which is providing a framework for encryption and authentications of your stream.
SRTP will be also use between your MGCP gateway and your IP Phone but you need to know that your SRTP keys are exchanged in cleartext session between the MGCP gateway and the Call Manager.
VXML is a W3C standard that allows voice-based interaction between human-users and computers applications. VXML can be used for applications and systems such as Auto-Attendant, voicemail or IVR, with VXML scripts performing functions such as playing prompts, collecting user input (DTMF and speech) and routing calls. VXML scripts can perform IVR functions similar to TCL scripts, the major difference is that whereas TCL scripts are usually device memory resident or downloadable from a TFTP Server. VXML scripts are usually interpreted by a voice browser after they are downloaded from a web server using http request (client/server model)
CRS is supporting VXML 2.0 applications
To activate the speech recognition on your IPCC server, you must use the following subsystem , a Media Resource Control Protocol (MRCP) Automated Speech Recognition (ASR) ( It is the client component).
If it comes with a separate ASR server like Nuance, you are able to enable speech recognition. So let’s review a little the conversation between the ASR server and IPCC server using MRCP.
MRCP is a mechanism which allows a client device (IPCC/phones) that requires audio stream processing to control processing resources such as ASR ad TTS servers ( for speech recognition and Text-to-speech conversion). Remind also that if you want to enable also Text-to-speech , you need to have another MRCP dedicated for TTS . So one MRCP ASR and one MRCP TTS.
MRCP relies on the Real Time Streaming protocol (RTSP) or SIP as a control protocol for setting up and controlling sessions. RSTP/SIP is also responsible for setting up media streams between the client and the server by using RTP (it is a kind of H245 negotiations)