Turning your ViA into a Phone Hybrid and integrate with an announcer

Turning Your ViA Codec into a Phone Hybrid

“Jake’s Take”

Tech Tips from Tieline’s U.S. Codec Expert Jacob Daniluck

Turning Your ViA Codec into a Phone Hybrid

In this month’s installment of Jake’s Take, I would like to add to the discussion from last month’s issue of the Stream Line, where I took on the topic of High-Quality Remote Broadcasting from Multiple Locations. This month, I explore the idea of adding a phone line into the audio stream to receive a guest’s phone call and take it to air while working remotely.

While in a studio, these call-ins get handled by using your Studio Phone Hybrid. With this setup the studio, everything usually works fine, and you likely don’t run into anything except poor phone line audio quality. However, when it comes to working from home without the studio hybrid, this creates a challenge for the engineer. He or she will need to re-route the audio through the studio down the return codec channel or give the remote user direct access to the phone line. When using your codecs return channel, you’ll end up increasing the delay between your host and the caller. With the use of the SIP Protocol on your audio codec, you can help reduce the number of components employed as well as the potential for audio delays between your host and caller.

Setting Up a SIP Trunk 

The first bit of the setup involves a SIP Trunk (‘Virtual’ Phone line) and the SIP PBX system used with your Studio Hybrid. You will need to make sure that there are two extensions configured on your SIP PBX. The first extension is for the Tieline ViA (Ext. A), while the second extension (Ext. B) is for a SIP Phone for the call screener. When setting up the call routing, you need to make sure that the ViA extension [A] doesn’t ring for inbound calls from the SIP Trunk. Instead, this extension is for internal call routing only (i.e., a transfer from another SIP Phone). The SIP extension [B] should be configured to receive inbound calls from your SIP Trunk and have access to transfer to other extensions.

Turning your ViA codec into a Phone Hybrid and integrate with an announcer
Setting up a SIP trunk and mixing it with announcer audio at a remote site.

 

The next step is to configure each device to connect to your studios SIP PBX. For the SIP phone, consult with your phone manufacture to program connectivity with the SIP PBX. On the ViA, you will program the SIP account and interface through the front panel or the Web Interface. Once configured, you will need to set up the specific interface for the ViA to use for the SIP connection as there are multiple transportation options.

Adjusting the Audio Matrix on the ViA

You can adjust the Audio matrix on the ViA codec directly using the touchscreen under Media settings, or using the WebGUI. When streaming live you have to modify the audio matrix and build a unique streaming program for the ViA. This program type used is a point-to-multipoint configuration (i.e., Stereo + IFB, Dual Mono, Triple Mono).

ViA-Dual-Mono-SIP-call-integration
Matrix edits in orange to support this use case with the default dual mono program on a ViA codec

In this Dual Mono program example, the first stream is for the main program audio (Encoder and Decoder 1), while the second mono stream is for the SIP Connection (Encoder 2 and Decoder 2). The act of having the SIP account linked to a specific audio stream has the caller appear on a particular encoder/decoder, just like having multiple hosts from multiple locations. The audio from the incoming PBX SIP call Ext. A is mixed with announcer audio and streamed back to the studio using Encoder 1.

It is now time to test connectivity with the SIP PBX. First, dial the station’s call-in line using an outside phone line. Once dialed, the SIP phone (Ext. B) should start to ring, and you should be able to pick up and screen the call. Once the caller is screened, you can pass the call onto the ViA Ext (Ext. A) by using a simple transfer. Once transferred, the ViA will auto pick up the call and place them on-air with your host.

Scenario 2:

ViA Dual Mono SIP call integration mix at studio
Matrix edits in orange to support mixing feeds at the studio using the dual mono program on a ViA codec

If you want greater control at the studio you can mix your remote announcer audio and incoming SIP PBX call Ext. B at the studio end. In this slightly modified scenario, the Ext. A SIP call is only used to monitor a talkback caller at the ViA codec and talk back to them. A clean feed of the announcer’s audio is streamed to the studio over Encoder 1 and this is mixed with a clean feed directly from PBX Ext. B at the studio. This scenario works best when IP latency between the studio and remote site is low. The following image displays the workflow in this scenario:

Turning your ViA into a Phone Hybrid
Configuring a SIP trunk and mixing downstream at the studio

 

Challenging conditions make it hard to have the same flexibility that existed in the main studio before being forced to broadcast remotely. With just a little bit of modification to existing equipment like your SIP PBX and ViA, you can expand your remote broadcasting ability and secure listener engagement even while on the road. For more information on the Tieline ViA please visit www.tieline.com/via

Request a Future Topic for Jake’s Take…

For those who are interested in sharing your story, or if you have an idea for a future “Jake’s Take”, please feel free to contact me directly at Jacob@tieline.com

Contact Tieline:

• For USA, Canada & Latin America contact: sales@tieline.com
• For Australia and International: info@tieline.com

(Jake’s Take on Turning Your ViA Codec into a Phone Hybrid , published 5th June, 2020)

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