Product Manual
Comrex EarShot IFB Manual I. Comrex Introduction 4 II. EarShot IFB 5 Introduction 5 About Mix-Minus 6 III. Setting Up EarShot IFB 7 Hardware Attachments and Connections 7 Audio Connections 8 Contact Closures 8 Making Connections 8 IV. Setting up EarShot IFB IP Information 10 Introduction 10 Configuring EarShot IFB IP Information 10 V. Operation of EarShot IFB 15 VI.
VII. System Behavior 24 Audio Settings 24 IFB settings 25 Other Options 26 PINs 27 VIII. More About SIP NAT Router issues with SIP IX. Why do I Hear Hear Myself Myself? Mix-Minus and Eliminating echo X. Information for IT Managers About EarShot IFB Mainframe XI.
I. Comrex Introduction About Comrex Comrex has been building reliable, high quality broadcast equipment since 1961. Our products are used daily in every part of the world by networks, stations and program producers. Every product we manufacture has been carefully designed to function flawlessly, under the harshest conditions, over many years of use. Each unit we ship has been individually and thoroughly tested. Comrex stands behind its products.
II. EarShot IFB Introduction Comrex EarShot IFB is a hardware-based system designed to answer phone calls and deliver live audio feeds to callers. EarShot IFB is designed to provide telephone-based live studio program and IFB audio to fieldbased remote broadcasts, like TV ENG reports. EarShot IFB has four audio inputs (or feeds) and callers can choose between these four feeds. Feed choice is made by callers using DTMF (Touch Tone) selection via PIN code.
EarShot IFB does not offer studio-side audio output functionality. No output ports to hear callers are provided. By default, EarShot IFB is a “one-to-many” device that sends only its own feed audio to callers. Callers can, via DTMF, enable their source audio to be added to the mix (to send to other callers on that feed). This creates a conferencing function. Also by default, incoming callers are placed into a “waiting room” until they input a PIN for a specific feed.
III. Setting Up EarShot IFB Hardware Attachments and Connections The figure below shows the rear panel of the EarShot IFB mainframe: 1 2 3 4 5 6 7 8 9 10 GPIO AES3 ANALOG USB PRI SEC SERIAL 11 VGA 12 1 Mains Power - Apply universal mains power (110-240VAC) to the IEC connector here. 2 ANALOG/AES3 Input Switch - This switch determines whether the leftmost XLR connector is used for analog FEED/IFB1 audio to callers, or is configured as an AES3 digital audio input.
Audio Connections All analog XLR audio inputs have a nominal level of 0dBu (full scale +20dBu). AES3 inputs support 32, 44.1, and 48KHz sampling rates. AES3 inputs are only provided for FEED/IFB ports 1 and 2. Analog input pinouts: AES3 input pinouts: Pin 1 Ground Pin 1 Ground Pin 2 Balanced audio + Pin 2 Data + Pin 3 Balanced audio - Pin 3 Data - Contact Closures Contact closure signals are available via the male 9-pin D connector on the back of the EarShot IFB.
In most ways, EarShot IFB will look like an ordinary computer to this network. In fact, EarShot IFB contains an embedded computer with a Linux-based operating system and a full network protocol stack. EarShot IFB is perfectly capable of working over most LANs, but there may be situations where a LAN is heavily firewalled, subject to overloaded traffic conditions, or has security concerns. Better performance is possible if EarShot IFB has its own Internet connection.
IV. Setting up EarShot IFB IP Information Introduction EarShot IFB features two Ethernet ports (Primary and Secondary), which provide the following functionality to the system: • Voice-over-IP service provider accounts • Attachment to PBXs • Attachment to gateway devices (bridges to POTS, ISDN, E1/T1, etc.
You will choose one of the possible configurations for each of the Ethernet ports.
Device Manager Install Device Manager on a computer on the same physical LAN as the EarShot IFB. (This is important, because Device Manager can generally only access devices that are on the same network it is using.) Once installed and run, press the “Scan for Devices” button in the upper left corner to scan your network for Comrex products. The results should look like this: There are two ways to set/reset the static IP information on the EarShot IFB: using Device Manager or utilizing Toolbox.
Toolbox The normal mode of changing any settings in EarShot IFB is by clicking the “Web Configuration” tab in the right pane of Device Manager. This opens the unit’s Toolbox interface. Toolbox is a web-based interface that allows you to set all relevant parameters of the product. Once you know the unit’s IP address, Toolbox can also be accessed via a browser by going to http:///cfg. Toolbox requires that you log on with a username and password.
Change the mode to Static, and input the IP address, Netmask, Gateway and DNS information for your network. Then click “Apply IP Settings”. Please note: If you need to access Toolbox via Device Manager after this, you’ll need to click “Scan for Devices” and log in again. Alternatively, you can open a browser to the new IP address (http:///cfg) and log in that way. The Primary Network page is also where you can adjust the port for web and xml services.
V. Operation of Earshot IFB EarShot IFB only processes incoming calls. By default, callers are parked in a “waiting room”, and a prompt is sent instructing the caller to enter a PIN. The PIN is pre-set by the administrator, and on verification the caller is added to one of the four possible “feeds” based on the PIN. By default, callers can only hear the Feed and not each other. EarShot IFB has an optional IFB function.
VI. Telephone Connections Introduction to SIP SIP (Session Initialization Protocol) is the standard used by EarShot IFB to talk to virtual phone lines. These lines must be created in some way externally before they are “applied” to EarShot IFB. “Applying” SIP lines involves configuring the mainframe with certain information about the lines and the location of the server that delivers them.
Setting up a SIP provider or PBX SIP provider’s info is entered using Toolbox, a web-based config utility. In most arrangements, a SIP provider can deliver several telephone channels to you over a single account. When you subscribe to a SIP provider, make sure you understand how many simultaneous channels are provided with that account. These channels will act like a traditional telephone hunt group, so in the case of channel one being busy, calls will be routed to channel two, etc.
The next three fields are where you should enter your account credentials: Account Username, Account Password, and Server/Realm. These should have been provided to you by the SIP provider. Account username is usually the name at the start of any SIP URI assigned. (More on that later). E.g., username@sip_provider. Make sure to enter only the username part. The Server/Realm is the address of a registration server maintained by the provider. No web prefixes are needed here, just the address. E.g., sip.comrex.
- Under SIP Settings Provider Binding Port - This port is assigned by EarShot IFB based on the number of providers you have assigned. Unless required, you should leave the default setting as is. Proxy Address - Most providers use the same server address for incoming call registration and proxying SIP traffic. If your provider shows a different address for proxy, enter it here. Registration Proxy Address - Some providers require REGISTER messages be sent to a different server.
- Extra Settings - Destination Match- This field is normally left blank. If an expression is entered here, the system will attempt to match the text in this field with the incoming calls “Destination” field. Calls that don’t match will be rejected. This is required when working with some PBXs. This field can also be set as a “regular expression” for a dial plan, which is a more complex topic and can be handled by Comrex support. As an example, entering the following value in the field: ^(1\d{3}555210\d)$ W
The previous figure shows the settings for gateways (Line Configuration > VoIP Providers > Add Providers > SIP Gateway Device). Many of the settings are populated automatically by EarShot IFB, but can be changed to any value you wish. - General Settings Name - Give your gateway a unique name. - Account Information Username and Password - Locally generated values that the gateway will use to register to EarShot IFB. - Sip Settings Address - The IP address of the gateway.
You may also come across certain PBXs that must deliver their extensions in this way. In order for EarShot IFB to receive these extensions, a SIP trunk must be created instead of a provider account. SIP Trunks also differ from normal SIP providers in several ways. Rather than having EarShot IFB “pull” the SIP channel from a provider, a SIP trunking provider will “push” the channel to a specific IP address of the user.
This would match the sequence “1xxx555210x” where “x” is any digit. Note, the Destination Match field is not used for SIP Trunks. This is used for registered providers only. Putting information in this field for SIP Trunks may result in call failure. Once created, the status of your SIP trunks will always show as “registered”. In the case of SIP trunks, this status is meaningless, since no actual data has been received from the provider until a call is established.
VII. System Behavior Back at the main Toolbox page, selecting System Behavior allows you to set how the EarShot IFB reacts to incoming calls. The options are divided into Audio, IFB, Other, and Line PINs settings. Audio Settings PIN-Free Mode - In normal operation, incoming callers are parked into the system’s “waiting room” and prompted for a PIN number to select their feed. PIN-Free mode instead parks all callers into Feed 4. It is still possible to change to other Feeds using DTMF control.
IFB settings IFB Feed Count - This function allows use of IFB channels. EarShot IFB has a total of four inputs, which by default are all “Feeds” that can be selected by callers. By engaging IFB, the user will sacrifice one or two of the Feed inputs, and use only the two or three remaining feeds. Only Feed input #1 and #2 can be changed into IFBs. This means that input #3 and #4 are always dedicated as normal feeds. A selection is provided to enable IFB.
Other Options Caller Feed Add PIN - DTMF sequence that will add a caller’s microphone audio to the feed when dialed. Caller Feed Remove PIN - DTMF sequence that will remove a caller’s microphone audio to the feed when dialed. Contact Closures - By default, the contact closure outputs are active when any call is active on a particular feed (e.g. a call active on Feed #1 triggers Contact Closure #1). The contact closure for each feed can be changed in the Contact Closure menu.
Next, press Start Trace. When you would like to end the capture, simply select Stop Trace. Once pressed, a new button titled Download Trace becomes available. Press this to download the packet capture file (a .pcap extension). This will start an http download to your browser. To review this file, we recommend using Wireshark, a free and open source packet analyser available online. PINs Callers will select their feeds by inputting DTMF (Touch Tones). The PINs used to select each feed are programmable here.
VIII. More About SIP Before we get into a conversation about routers and SIP issues, it’s important to understand the basics of how SIP works. The great thing about SIP is that the same protocol is used between devices along the chain. In our example shown in Figure the figure above, a SIP PBX uses the protocol to register and place and receive calls with a SIP service provider on the Internet. In this scenario, the provider is acting like a SIP host and the PBX is acting like a client.
Technically, as shown above, a SIP client opens one channel to its host when it registers. This channel is kept open indefinitely, and uses the TCP transport layer for reliability. This channel is used by the SIP client to request an outgoing call, as well as by the SIP host to notify the client of incoming calls. The channel is initiated from the client and kept open with keep-alive data so it doesn’t have any issue when running through routers that incorporate Network Address Translation (NAT).
Issue #2 is more complex, but there are some workarounds: 1 The simplest way of avoiding NAT issues is not to use NAT at all. If your router supports a DMZ function, this will have the effect of putting your client (EarShot IFB) open to all incoming traffic on the Internet. Alternately, EarShot IFB can be located directly on an open Internet connection with a fixed, public IP programmed into it. 2 Many modern routers and firewalls have a feature called SIP ALG that may be either on or off by default.
IX. Why do I Hear Hear Myself Myself? Mix-Minus and Eliminating echo Studio telephone integration is a two-way process. The caller must send his audio to the studio, but also receive a return feed that allows him to interact with other sources, like a host. An important element of voice telephony involves allowing a speaking party to hear his own voice in his own earpiece. This sidetone provides a speaker the comfort of knowing his voice is getting through, and makes two-way communication flow more easily.
Even in “old-fashioned” analog telephone circuits, it’s possible to create a 10mS round-trip delay on a long distance call. Now add in the requirement that modern VoIP-based systems have inherent windowing and buffer delays, and its easy to pile up over 100mS round-trip on a call. A delay of this length will typically not impede interactive conversation, but will certainly create an intolerable “slapback” environment if the caller hears his own voice delayed.
SINGLE MIX-MINUS NO TELEPHONE FEEDS In some environments, it’s only important that the caller hear the in-studio host, and less urgent that the onair caller be able to hear automation, news reports, codecs etc. It’s possible that simply routing an amplified version of the studio microphone signal to the “send” input will meet those needs. Some studio telephone systems, like Comrex EarShot IFB, allow telephone callers to appear on one of two outputs (and therefore on two, separate console faders).
TWO MIX-MINUS FEEDS 34
X. Information for IT Managers About EarShot IFB IFB Mainframe EarShot IFB mainframe is an embedded Linux-based device with dual 10/100/1000Base-T Ethernet ports. The device contains an optimized version of the Linux kernel (at this writing, 3.12). The IP parameters are set using a GUI that requires attachment of a keyboard and VGA monitor to the device. Alternately, during the first five minutes of power up, the IP parameters may be set by a PC on the local LAN using a proprietary broadcast UDP protocol.
Outgoing Services As described above, the mainframe will make outgoing connections to register with SIP providers (usually to port UDP 5060) combined with incoming and outgoing RTP in the range of UDP 16384-16432. An NTP client is implemented by default to UDP 123. The mainframe syncs by default to a Comrex cloud server at TCP 8090. This is currently not required for normal operation. SIP connection attempts will sometimes make external STUN connections to UDP 3478.
XI. Software Licenses EarShot IFB contains both proprietary and open source software. All proprietary portions are Copryight © 2012 Comrex Corporation, all rights reserved. COMREX CORPORATION END USER LICENSE AGREEMENT IMPORTANT READ CAREFULLY: This End User License Agreement (Agreement) is a legal and binding agreement between you (either an individual or a single entity) and Comrex Corporation (Comrex) for the software contained within this product.
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