webrtc|ICTBroadcast

WebRTC (Web Real-Time Communications)

Tahir Almas — Tue, 29 Apr 2025 07:26:17 +0000

ICTBroadcast introduces another impressive call center tool that automates the business processes of call centers by providing a WebRTC based Agent panel.

WebRTC (Web Real-Time Communications)

It is cutting-edge technology and norm that is very exciting, strong, and extremely innovative. WebRTC leverages a range of plugin-free APIs that can be used in desktop and mobile browsers alike, and all popular and modern browser companies are increasingly being supported.
WebRTC uses different specifications and protocols, most of which will be covered in this article. Which include data streams, signaling, JSEP, STUN/TURN servers, ICE, SIP, NAT, SDP, UDP/TCP, sockets for the network, etc.

WebRTC use for Calling / Communication directly from Browser

WebRTC is contain HTML5 specification. It allows browsers to communicate with one another directly and send audio, video, and data in real-time. It is used by services like Facebook Messenger, Slack, Houseparty, Webex, and now Skype to function from browsers and, in many cases, in their backend infrastructures for media production.

WebRTC in ICTBroadcast

ICTBroadcast presenting new and rich in functionality WebPhone based on WebRTC. This new WebPhone in ICTBroadcast allows agents to accept calls without any external hardware or software via their browsers. More about the new WebRTC release in ICTBroadcast 4.1 given below.

How to Use new WebPhone (WebRTC) in ICTBroadcast

To access the new WebPhone is simple as before, Just log in as an agent by clicking “Agent Login” as showing in the image.
After clicking agent form will appear for an agent to log in, just provide the agent creds, and the agent will be in the new WebPhone.

Beautiful and User-Friendly Interface

ICTBroadcast team design the new WebPhone super simple, well design and user-friendly UI. Where all the necessary information is showing in one screen view. let’s know more about it.

WebPhone in Agent Dashboard

New WebPhone UI attached on the right side of the agent dashboard to get rid of the trouble of dragging and dropping to left, right, up, or down.

What’s new in new WebPhone

There are multiple new features that have been added, let’s highlight them.

Agent Status Predictive/Manual

This option allows the agent to set his/her status “Manual” or “Predictive“. When the agent selects the Predictive option, It will let the decide ICTBroadcast to chose to dial the next contact from the list, and in the Manual option, Agent will have to that by him/her self.

Show Script

During the call session, Agent can easily see the script by clicking on “Show Script“

Call Transfer

While calling agent can transfer the current call as he/she needed to the desired agent. When the agent clicks on the “Transfer” button all agents will show in the list and the call will be transfer simply by a click.

DTMF/NumPad

Useful Numpad provided along with new WebPhone. The following is a list of usage-able main sequences that the agent will use for various purposes e.g.
“0” For Hangup.
Press #1 for a call transfer.

Contact List Tab

Added a new tab named “Contact List”. Where the agent can see all contacts list of default selected campaigns. There Agent can see additional information like Name, Phone, Email, Schedule, Assigned, and 2 operations buttons, one for “Remove” to remove the contact and one for “Dial” the contact for each.

Script Tab

In this tab, An agent can view script/content that is used in the campaign.

Update Tab

In this tab section, The agent can see all the information about contact during the call and can update the contact as well.

Disposition Tab

In this tab, the agent can re-schedule the call if the recipient does not attend the call or powered off. An agent can reschedule it by pre-defined scheduling option or he/she can make a custom schedule by selecting the “Custom Schedule” option.

Agent History

At the bottom of the new WebPhone, there is a new section to show the agent history such as Date and Time along with Name, Phone, Campaign name, Queue Wait and Handle Time

The post WebRTC (Web Real-Time Communications) first appeared on ICTBroadcast.

Leveraging WebRTC for Browser-Based Communications in ICTBroadcast

Tahir Almas — Sat, 26 Apr 2025 08:01:22 +0000

ICTBroadcast is an Ai-powered auto dialer software and unified communications platform that offers companies robust voice, SMS, email, and fax communication options. ICTBroadcast allows for seamless browser-based communication for voice, video, and data sharing thanks to the incorporation of WebRTC (Web Real-Time Communication). WebRTC streamlines communication procedures by eliminating the requirement for external hardware, software, or plugins. WebRTC improves user comfort and accessibility by allowing real-time interactions straight from a web browser, whether for inbound customer care or outbound marketing campaigns.

ICTBroadcast enables companies to simplify their operations and expand their communications with ease by using WebRTC. The integration enhances efficiency without requiring any extra infrastructure or setup by enabling remote teams and call centers to handle communications via an easy web interface. The solution offers secure, top-notch connections for businesses of all sizes, regardless of their size. ICTBroadcast transforms how businesses interact with customers by using WebRTC to enable real-time, browser-based communication.

What does WebRTC mean in ICTBroadcast?

WebRTC is an open-source technology that allows users to share audio, video, and data in real time directly through their web browsers. It enables smooth peer-to-peer communication without the need for extra plugins or software installations. WebRTC is a key facilitator of browser-based communications in ICTBroadcast, offering businesses a flexible and affordable way to enhance customer interaction.

With ICTBroadcast’s WebRTC, users may engage in high-quality voice and video conversations directly from their browsers, providing a seamless and intuitive experience. WebRTC in ictbroadcast improves its platform’s features, providing a scalable answer that aids enterprises in managing communications, delivering customer support, and running effective marketing campaigns without the need for complicated infrastructure. The technology enables real-time communication for both agents and customers, removing the need for external devices like SIP hardware or softphones.

How WebRTC is Used by ICTBroadcast

ICTBroadcast transforms how businesses manage voice, video, and data interactions by using WebRTC technology to provide smooth, browser-based communications. WebRTC integration allows marketing departments, customer support teams, and call centers to communicate in real-time directly through their web browsers without the need for any extra software or equipment. This integration greatly streamlines the user experience by facilitating easy connections between agents and customers, which improves communication.

The strength of WebRTC in ICTBroadcast comes from its capacity to facilitate trustworthy, secure, and high-quality interactions without the need for complex setups. Users may handle incoming support, conduct video conferences, launch marketing campaigns, and make calls—all from a single web interface. This minimizes both hardware and maintenance expenses by removing the necessity for distinct telephony equipment or softphones. Additionally, WebRTC’s low-latency connection enables companies to depend on real-time interactions, which are essential for enhancing operational effectiveness and customer satisfaction.

Key Features of WebRTC Integration in ICTBroadcast

Voice Broadcasting (Automated Call Campaigns)

ICTBroadcast incorporates WebRTC to allow for smooth voice broadcasting campaigns that can be launched straight from the browser. Companies may quickly start automated outbound calls to thousands of people in real-time, delivering either interactive prompts or pre-recorded messages. By saving time and labor, this function enables organizations to successfully conduct product promotions, political surveys, marketing campaigns, and emergency alerts. For maximum impact when reaching a big audience, the WebRTC integration guarantees crystal-clear, uninterrupted audio quality.

SMS Broadcasting to Increase Reach

With its sophisticated SMS broadcasting capability, ICTBroadcast enables organizations to send mass messages straight from the web interface using WebRTC. This is especially helpful for alerts, event reminders, special promotions, and customer interaction. WebRTC simplifies the process, allowing companies to communicate with thousands of customers. ICTBroadcast provides reliability and speed by leveraging WebRTC, guaranteeing that all messages arrive at their intended recipients without delay.

Customer Support in Real Time (Incoming Calls)

By allowing high-quality voice and video conversations through any web browser, WebRTC improves ICTBroadcast’s real-time customer support. Customers can instantly connect with support agents when they require help, which enhances both service quality and efficiency. Whether dealing with complaints, questions, or technical issues, support agents may quickly resolve problems thanks to WebRTC’s low-latency communication. With this capability, companies may provide better customer support that promotes satisfaction and loyalty among their clients.

Integration of Interactive Voice Response (IVR)

ICTBroadcast integrates effortlessly with Interactive Voice Response (IVR) systems using WebRTC. This enables companies to automate call handling, facilitating customers’ menu navigation, department selection, and self-service activities like troubleshooting common issues or checking balances. By providing a higher degree of customer interaction and lowering the load on support personnel, the WebRTC feature enhances operational efficiency by making these interactions more fluid and responsive.

Collaborative Calling and Conferencing

ICTBroadcast makes it simple for organizations to hold team meetings, sales calls, or customer consultations by using WebRTC to provide conference calling straight through a browser. This function enables real-time interaction between participants through voice and video conferencing, irrespective of their geographic location. WebRTC is perfect for remote team management, brainstorming sessions, and collaborative work since it keeps the connection clear and stable even with many users. Businesses can utilize this function to enhance communication among remote teams, as well as for training and team building.

Campaign Reporting and Analytics

The WebRTC integration in ICTBroadcast offers more than just communication; it also offers real-time reporting and analytics. From the web interface, companies may evaluate agent performance, track success rates, analyze call data, and monitor campaign success. This functionality enables companies to enhance overall campaign results, refine their marketing tactics, and make choices based on data. Businesses may strategically change their activities in real-time by looking at live data, which improves efficiency and return on investment.

Click-to-Call Features

WebRTC’s click-to-call function allows visitors to websites to immediately reach agents by clicking a simple link or button. Customers may make voice or video calls straight from their browser by clicking the button, eliminating the need to manually dial phone numbers. This immediate interaction improves user satisfaction, decreases response times, and raises the chance of conversions. Businesses may provide this smooth, interactive communication experience on their websites thanks to ICTBroadcast’s integration with WebRTC.

End-to-End Encrypted Secure Communication

Companies that use ICTBroadcast prioritize security above all else. WebRTC protects sensitive customer information and communications by ensuring that all voice, video, and data transmissions are encrypted with end-to-end security. This feature is especially important for fields like healthcare, finance, and legal services, where confidentiality is essential. ICTBroadcast guarantees that companies may offer secure and private communications by integrating WebRTC, which helps build customer trust and adhere to regulatory requirements such as HIPAA or GDPR.

Scalable VoIP Alternatives

The WebRTC integration in ICTBroadcast enables scalable VoIP solutions, permitting companies to grow their communication infrastructures as required. WebRTC guarantees that the platform can accommodate a rising number of agents, clients, and calls without losing quality, regardless of whether it is used for large business settings or small-scale operations. ICTBroadcast is a flexible and future-proof answer for any kind of communication needs because of its scalability, which allows companies to adapt to shifting operational needs or rising demand.

Support for Multiple Channels of Communication

ICTBroadcast with WebRTC provides extensive multilingual communication support for voice, SMS, and email broadcasting, all of which may be controlled via a single web interface. This feature enables companies to connect with customers using the most appropriate channels, whether that be through voice campaigns, automated SMS messages, or email interactions. WebRTC guarantees that every communication channel runs seamlessly, providing companies with flexibility and control over their outreach methods, all while ensuring high-quality, real-time interactions.

Challenges with WebRTC in ICTBroadcast

Problems with Browser Compatibility: WebRTC might not function smoothly across all browsers, especially older ones, which could result in inconsistent user experiences for certain clients or agents.

Network Lag and Performance Fluctuation: Problems such as delays and poor quality on unstable internet connections can impact WebRTC performance, which can be detrimental to real-time assistance or large-scale campaigns.

Security Threats in Open Networks: Although WebRTC provides encryption, using it over unprotected networks, such as public Wi-Fi, can expose communications to possible security threats.

Integration Complexity with Current Systems: It may be technically difficult to integrate WebRTC into current telephony or CRM systems within ICTBroadcast, and it may need further bespoke development.

Server Load and Scalability: Businesses may need to optimize their infrastructure for seamless WebRTC performance as a result of the substantial strain that high traffic and large-scale communication campaigns can place on servers.

Overcome Challenges with WebRTC in ICTBroadcast

Guaranteeing Browser Compatibility: ICTBroadcast can promote users to update their browsers to improve browser compatibility. A smoother experience across various devices can also be accomplished by implementing fallback choices or identifying compatibility concerns early on.

Lowering Network Latency and Enhancing Performance: By investing in high-quality internet connections and using Content Delivery Networks (CDNs) to lower latency, businesses may improve WebRTC performance. A more consistent experience is provided by adaptive bitrate technology, which adjusts call quality according to available network bandwidth.

Improving Protection on Public Networks: To reduce security threats, companies can use VPNs or other secure connection techniques when using WebRTC over public networks. WebRTC encryption can be combined with extra security layers like TLS or SRTP to keep sensitive communications safe.

Making Integration with Current Systems Easier: Businesses can make it easier to integrate WebRTC with ICTBroadcast by using API-based solutions or getting help from developers who are familiar with both WebRTC and ICTBroadcast systems. This speeds up the deployment process and lowers integration complexity.

Improving Server Infrastructure for Growth: By ensuring that the infrastructure is elastic and cloud-based and by improving server load balancing, ICTBroadcast can address scalability problems. This ensures that the system maintains steady performance even during peak usage times by enabling it to automatically increase or decrease depending on real-time demand.

Advantages and Disadvantages of WebRTC in ICTBroadcast

Advantages:

Affordable Communication: By removing traditional telephony systems and infrastructure, WebRTC saves expenses and enables companies to conduct communication initiatives online.

Improved Customer Interaction: By facilitating real-time voice, video, and messaging, it enhances engagement and customer satisfaction in marketing and support campaigns.

Simple Browser-Based Access: WebRTC enables immediate communication for both agents and clients without the need for extra software and with smooth access.

Cons

Dependency on the Internet: WebRTC depends on internet quality, and bad network conditions can result in distorted audio or dropped calls.

Compatibility Problems: WebRTC may not be fully supported in some older browsers or less frequently used systems, which could lead to access problems for some users.

Management of Security: Although WebRTC provides encryption, further safeguards are needed to secure communications and prevent data breaches, particularly over public networks.

Frequently Asked Questions

What does WebRTC stand for in ICTBroadcast?

WebRTC, or Web Real-Time Communication, is a technology that allows for real-time voice, video, and messaging interaction directly through web browsers. It is integrated into ICTBroadcast.

What are the benefits of WebRTC for companies using ICTBroadcast?

WebRTC offers a low-cost, browser-based option that removes the requirement for extra hardware, improving customer interaction and simplifying communication campaigns.

In ICTBroadcast, WebRTC enhances communication by enabling real-time audio and video interactions between users. This facilitates more efficient collaboration and communication, lowers latency, and enables instant media sharing. WebRTC also supports interoperability with various devices and platforms, making it easier for users to connect and communicate. WebRTC improves communication by allowing voice, video, and SMS interactions to take place directly through the browser, which enhances both the effectiveness of customer engagement and support.

Which browsers in ICTBroadcast support WebRTC?

Most current browsers, including Google Chrome, Mozilla Firefox, Safari, and Microsoft Edge, support WebRTC. Older or less popular browsers might not be entirely compatible.

How secure is WebRTC in ICTBroadcast?

WebRTC offers integrated encryption for secure communications. However, especially over public networks, additional precautions like utilizing VPNs or other security protocols may be required to guarantee total security.

The post Leveraging WebRTC for Browser-Based Communications in ICTBroadcast first appeared on ICTBroadcast.

Integrating WebRTC with SIP: A Complete Guide for 2026

Tahir Almas — Fri, 18 Apr 2025 06:09:31 +0000

Quick answer: WebRTC-to-SIP integration lets browsers join the same call paths as traditional SIP softphones and PBX endpoints by translating WebRTC signaling (ICE, DTLS-SRTP, JSEP) into standard SIP/RTP at a gateway. The gateway runs in front of your SIP server (FreeSWITCH, Asterisk, Kamailio), handles codec transcoding when needed, and exposes the same call legs to recording, IVR, queues, and downstream business logic. Done right, you get a no-install browser softphone with full PBX feature parity.

WebRTC turned every modern browser into a real-time communication endpoint. SIP is still the protocol holding the global voice network together. Most production unified communications stacks now run both, and the gateway between them is the piece that makes a browser-based softphone feel exactly like a desk phone to the back-end PBX.

This guide walks through what the integration actually does, how to architect it for production, the security and scaling concerns that matter, and where it fits in 2026 deployments.

WebRTC and SIP: What Each One Does

The two protocols solve different problems and only partially overlap.

	WebRTC	SIP
Where it runs	Browser, mobile WebView, native WebRTC libraries	Hard phones, softphones, PBX, SBC, carrier networks
Signaling	JSEP (offer/answer over any transport , typically WebSocket)	SIP messages over UDP, TCP, or TLS
Media	SRTP with DTLS key exchange (mandatory)	RTP, optionally SRTP
NAT traversal	ICE with STUN and TURN built in	SIP-aware NAT traversal, often via SBC
Codec set	Opus and VP8/VP9/AV1 standard	G.711, G.729, Opus, video varies by endpoint
Discovery	None , application-defined	SIP REGISTER and Location service

WebRTC was designed for peer-to-peer browser sessions with the application layer doing signaling. SIP was designed for call control across heterogeneous networks. The integration glues the strengths of each together.

Why Integrate WebRTC with SIP?

Browser-based softphones. No plugin, no install, no separate native app. Your agents log into a web app and start taking calls.
Click-to-call from any web property. Embed a call widget on a marketing site, a help center, or a customer portal that ends up in your existing call queues.
Same back-end logic. The PBX, IVR, recording, and CRM hooks you’ve already built keep working , the browser just looks like another SIP leg.
Lower endpoint cost. Browser softphones replace desk phones for most knowledge workers, and HR onboarding for remote staff drops to “open this URL.”
Modern audio quality. Opus is the WebRTC default, which beats G.711 and G.729 at the same bitrates and is now well-supported on the SIP side too.

How the Integration Actually Works

The Gateway Pattern

The integration centers on a WebRTC-to-SIP gateway. The gateway:

Accepts a WebSocket connection from the browser carrying SIP-over-WebSocket (RFC 7118) or a custom signaling protocol that the gateway translates to SIP.
Runs the ICE candidate exchange to establish the browser-side media path through STUN/TURN.
Negotiates DTLS-SRTP keys with the browser for encrypted media.
Opens a normal SIP call leg to the back-end PBX or SBC, transcoding codecs if the PBX doesn’t speak Opus.
Bridges the two legs in real time, applying SDP rewrites, codec transcoding, and any media policies (recording, transcription, DTMF detection).

SIP Server Configuration

FreeSWITCH, Asterisk, Kamailio, and ICTBroadcast all support WebRTC-to-SIP either natively (FreeSWITCH and Asterisk via their WebRTC profiles) or through a paired component (Kamailio with RTPEngine). The configuration always involves:

A WebSocket listener on TLS (wss://) for SIP signaling from browsers
An RTP/SRTP profile that supports DTLS key exchange
Codec ordering that prefers Opus when both sides support it, with G.711 as the safe fallback
Optional transcoding when the back-end SIP leg needs G.711 or G.729

Network Configuration

WebRTC’s ICE machinery handles NAT, but it needs help. In production:

Run a TURN server (coturn is the standard) reachable on UDP and TCP, with TLS for fallback through restrictive firewalls.
Open the right ranges. Browsers establish RTP on dynamic UDP ports , the firewall has to permit the gateway’s media range.
Front the WebSocket listener with a load balancer that supports WebSocket upgrade and sticky sessions.
Place the gateway and TURN server close to your users geographically to keep RTT low.

Security

WebRTC mandates DTLS-SRTP, so the browser-to-gateway leg is always encrypted. Match that on the SIP side:

Use TLS for SIP signaling and SRTP for media on the back-end leg
Authenticate WebSocket connections with short-lived tokens issued by your application server, not embedded SIP credentials
Rate-limit WebSocket and SIP REGISTER on the gateway to absorb credential-stuffing attempts
Run TURN with credentials that rotate, ideally tied to the same session tokens

Testing and Monitoring

Every WebRTC-SIP deployment needs three layers of observability:

Signaling. Capture SIP messages on the gateway and the back-end PBX. Wireshark, sngrep, and Homer all work.
Media. Track jitter, packet loss, and round-trip time per call. The browser exposes getStats() for the client side; the gateway exposes its own RTCP stats.
Quality scores. POLQA and MOS estimation on call recordings catch regressions early.

What WebRTC-SIP Integration Delivers in Practice

Browser-to-PBX Compatibility

A browser softphone shows up to your PBX as a regular SIP extension. Voicemail, hunt groups, queues, IVR, and recording all work without extra logic.

Audio and Video, Same Stack

WebRTC carries audio and video on the same connection. If your back-end SIP infrastructure handles video (most do via H.264 or VP8), browser endpoints can do video calls into the same call paths as desktop apps.

Data Channels

WebRTC’s data channel rides alongside the audio. Use it for in-call chat, screen-share signaling, presence updates, or real-time CRM event sync , all without extra connections.

Encryption by Default

DTLS-SRTP isn’t optional in WebRTC. Combined with TLS-secured SIP and SRTP on the back end, you get encrypted media end-to-end across the call.

Scalability

The gateway pattern scales horizontally. Add gateway nodes behind a load balancer, scale RTP-relay capacity (RTPEngine or the gateway’s own bridge), and the system handles thousands of concurrent browser legs per cluster.

Trends Worth Watching in 2026

WebTransport and WebCodecs. Lower-level browser APIs that complement WebRTC for specialized cases like ultra-low-latency streaming.
AV1 codec adoption. Better video quality at the same bitrate than VP9 , most modern browsers and PBX builds now negotiate it.
AI-assisted call processing. Live transcription, sentiment, and keyword spotting running on the gateway or in a sidecar service that reads from the bridged media stream.
WebRTC SFUs replacing MCUs. Selective forwarding for video meetings has fully replaced server-side mixing for everything except gateway interoperability.
Federated identity. SSO via OIDC for both the WebRTC side and SIP authentication backbones is now standard rather than bespoke.

Common Challenges

NAT and Firewall Traversal

The single most common failure mode. Symptom: signaling works, audio doesn’t. Fix: deploy TURN with TLS fallback, verify the gateway’s external IP advertised in SDP, and test from networks that block UDP.

Codec Mismatch

Browsers default to Opus. Older PBX endpoints may only do G.711 or G.729. Either transcode at the gateway (CPU cost) or force Opus end-to-end where you control both sides.

Latency and Jitter

Geographic distance between user and gateway matters. Keep the gateway close to users, monitor RTT continuously, and use adaptive jitter buffers on both sides.

Browser Inconsistencies

WebRTC is standardized, but browsers ship at different paces. Test on Chrome, Firefox, Safari, and Edge, and watch the release notes for breaking changes to media APIs.

Authentication and Session Management

Don’t ship SIP credentials to the browser. Issue short-lived tokens from your app server, validate them at the gateway, and rotate aggressively.

Where WebRTC-SIP Integration Pays Off

Contact centers. Browser-based agent desktops with no installs, click-to-call from CRM records, and full PBX integration on the back end.
Click-to-call on websites. A button on the marketing site that drops the visitor into a queue without leaving the page.
Telehealth. Patient-side video appointments in a browser, provider-side integration with the clinic’s existing phone infrastructure.
Distance education. Browser-based virtual classrooms with PSTN dial-in for parents and remote students.
Customer support. Live chat that can escalate to a voice or video call inside the same browser session.
Internal collaboration. Chat, meet, and call from one web app instead of three separate tools.

Trade-Offs to Be Aware Of

What You Gain

No-install endpoints across every device with a modern browser
Encrypted media by default
Flat economics , pay for infrastructure, not per-seat licenses
Tight integration with web applications and CRMs

What You Pay For

Higher operational complexity than a pure SIP deployment , TURN, gateway, and signaling all need monitoring
Browser-quirk testing , every release cycle introduces small WebRTC behavior changes
CPU cost for transcoding when codecs don’t line up
A learning curve for teams who’ve never run WebRTC infrastructure

Frequently Asked Questions

Do I need a dedicated WebRTC-SIP gateway?

Yes, in some form. Either a dedicated product (Janus, RTPEngine + Kamailio, FreeSWITCH with mod_verto) or the WebRTC profile built into your existing PBX. The browser side and the back-end SIP side don’t speak directly to each other.

Can I use ICTBroadcast as the SIP back-end for browser softphones?

Yes. ICTBroadcast is built on FreeSWITCH, which has mature WebRTC support. Configure the WebRTC profile, point your browser softphone at the WebSocket endpoint, and the browser leg behaves like any other SIP extension to the platform.

What does TURN actually do?

TURN relays media when direct peer-to-peer paths are blocked by NAT or firewalls. ICE tries direct paths first; if they fail, it falls back to TURN. Without a TURN server, a meaningful percentage of users will see signaling succeed but audio fail.

Is browser audio quality really as good as a desk phone?

Generally better, when network conditions allow. Opus at 24-48 kbps outperforms G.711 at 64 kbps and crushes G.729. The variable is network , desk phones on wired ethernet often beat browsers on flaky Wi-Fi, but the codec itself is no longer the limiting factor.

How many concurrent browser calls can one gateway handle?

Without transcoding, a modern gateway server handles a few thousand concurrent browser legs. With transcoding (Opus to G.711, for example), expect that to drop to the low hundreds per server. Scale horizontally behind a load balancer if you need more.

What about mobile browsers?

Modern mobile browsers (Chrome, Safari, Firefox on iOS and Android) all support WebRTC. The same gateway and configuration that serves desktop browsers serves mobile too, with the usual caveats about network quality and battery.

The Bottom Line

WebRTC-to-SIP integration is the bridge between modern browsers and the global SIP infrastructure that still routes most business voice. The architecture is well-understood, the open source pieces are mature, and the operational pattern (gateway in front of your SIP server, TURN for NAT, TLS everywhere) is proven across thousands of production deployments.

If you’re building a contact center, a click-to-call experience, telehealth, or any browser-first communication app and want it to land cleanly inside an existing PBX or carrier network, this is the pattern to build on. ICTBroadcast ships with WebRTC support out of the box, and ICT Innovations has been deploying WebRTC-SIP stacks since the standard stabilized.

Need a hand scoping or deploying it? Open a ticket at service.ictinnovations.com and we’ll walk through the topology and capacity planning with your team.

The post Integrating WebRTC with SIP: A Complete Guide for 2026 first appeared on ICTBroadcast.

Testing Press 1 Campaign / IVR with WEBRTC based webphone / Call Center module

Tahir Almas — Wed, 13 Apr 2022 07:12:05 +0000

⚠️ Note:
Please configure Gateway / SIP Provider account, add Contact, and Voice Recording before going through these instructions.

1. Create Agent through Admin / User Portal

Visit: Admin Login
Login as Admin and create an extension:

Path: Administration Menu ➔ Agent / Extensions ➔ “New Extension” button
Action: Enter Agent Name, Number, and Password, then click Save.

2. Create Queue / Skill Group

Path: Administration Menu ➔ Agent / Extensions ➔ “New Extension” button
Action: Enter details, select “Skill Group” from the “Forward To” dropdown, select Agents for this Queue, and click Save.

3. Assign Extension and Queue to User

Path: Administration Menu ➔ User Management
Action: Click the drop-down arrow under “OPERATIONS” for any user. Select agents and skill-groups, then click Assign.

4. Agent Portal Steps

Login as Agent from a different computer: Agent Login
Enter Agent Number and Password.
Click the “Ready” button from the agent phone portal.
Once ready, you will hear background music.

5. Launch Agent / Interactive Campaign

Go to Campaign Menu ➔ Campaign Management ➔ Add New.
Select Agent / Interactive (Press 1) campaign.
Enter Campaign Name, select Contact Group, and choose your Queue / Skill Group.
Click Save to launch.

Note: When a recipient presses the required key, the call is transferred to a live agent.

Using Webphone Features

Access the integrated webphone via the Agent Panel dropdown.

Set Agent Status

Change your status using the Status Button (Available or Leave).

Note: Only Available agents receive calls.

Call Transfer

Click the Call Transfer button and select the target extension.

Load Contact

Click “Load Contact” to view/edit prospect details within the ICTBroadcast internal records.

Keypad / DTMF Shortcuts

Action	Key Sequence
Hangup	*0
Call Transfer	#1 + Extension
Attended Transfer	#2 + Extension

How ICTBroadcast’s latest released agent panel will help you

Tahir Almas — Fri, 08 Oct 2021 08:42:09 +0000

ICTBroadcast introduces another impressive call center tool that automates the business processes of call centers by providing a WebRTC based Agent panel.

WebRTC (Web Real-Time Communications)

WebRTC use for Calling / Communication directly from Browser

WebRTC in ICTBroadcast

How to Use new WebPhone (WebRTC) in ICTBroadcast

To access the new WebPhone is simple as before, Just log in as an agent by clicking “Agent Login” as showing in the image.
After clicking the agent form will appear for an agent to log in, just provide the agent creds, and the agent will be in the new WebPhone.

Beautiful and User-Friendly Interface

ICTBroadcast team design the new WebPhone super simple, well design and user-friendly UI. Where all the necessary information is showing in one screen view. let’s know more about it.

WebPhone in Agent Dashboard

New WebPhone UI is attached on the right side of the agent dashboard to get rid of the trouble of dragging and dropping to left, right, up, or down.

What’s new in the new WebPhone

There are multiple new features that have been added, let’s highlight them.

Agent Status Predictive/Manual

Show Script

During the call session, Agent can easily see the script by clicking on “Show Script”

Call Transfer

While calling agent can transfer the current call as he/she needed to the desired agent. When the agent clicks on the “Transfer” button all agents will show in the list and the call will be transfer simply by a click.

DTMF/NumPad

Contact List Tab

Added a new tab named “Contact List“. Where the agent can see all contacts list of default selected campaigns. There Agent can see additional information like Name, Phone, Email, Schedule, Assigned, and 2 operations buttons, one for “Remove” to remove the contact and one for “Dial” the contact for each.

Script Tab

In this tab, An agent can view script/content that is used in the campaign.

Update Tab

In this tab section, The agent can see all the information about contact during the call and can update the contact as well.

Disposition Tab

Agent History

At the bottom of the new WebPhone, there is a new section to show the agent history such as Date and Time along with Name, Phone, Campaign name, Queue Wait and Handle Time

The post How ICTBroadcast’s latest released agent panel will help you first appeared on ICTBroadcast.

Set Up Call Queue in ICTBroadcast

Tahir Almas — Fri, 08 Oct 2021 08:31:06 +0000

To set up a queue for your call center, following are the steps:

Create Queue

Log in to ICTBroadcast as Admin or User
Navigate to the Main Menu → Administration
Select the Agent/Extension submenu and click to open
Click the “New Agent” button at the top-right corner
Select Agent Name and assign an Extension
Set Forward To by choosing the appropriate Skill Group
Click Save to complete the agent creation.

Optionally, refer to the Forwarding → Agent Module for additional forwarding settings

Assign extension

Once an extension has been created, you have to assign it to a user.

Select the sub-menu “Client Management” under the main menu Administration
A list of created Clients will be shown, choose the Clienr and click Assign Extension
The available extensions will be displayed here. Choose the extension and click Assign Selected

Create Campaign

Choose Campaign from menu
Click on Interactive Campaign
Fill all the fields, and in General Options, choose extension
Click Save

Queue Calls with custom Music

Call Queue in ICTBroadcast

Agent Module in ICTBroadcast

The post Set Up Call Queue in ICTBroadcast first appeared on ICTBroadcast.

ICTBroadcast Service Provider Edition Agent Guide (WebRTC)

Tahir Almas — Thu, 07 Oct 2021 12:08:19 +0000

ICT Broadcast brings another amazing feature for call centers and Agent support! WebRTC based Agent Panel for Agent to interact with callers / customers . Agent panel will supports Voice, SMS, Fax, and email communications technologies in future however we will discuss only voice communications here.

What is a WebPhone? and How does it help Agents?

A WebPhone is a WebRTC based phone that allows agents to receive incoming and make outbound calls using their browsers without any additional hardware or software. ICTBroadcast offers integrated WebPhone for Agents. It empowers agents to perform the following actions.

log in as an Agent

Users can even access integrated WebRTC Web Phone by login in as an agent by giving an extension and password.

Set Agent Status

ICTBroadcast provides an option to change the status of Agent, the agent can set his status as ready and not ready by selecting the appropriate status option from the Agent status box,

Note: The agent with ready status can receive, forward, or make calls

Break Reasons

When stepping away from their station, agents can set a named break reason rather than switching to a generic Not Ready status. ICTBroadcast supports three standard break reasons: Away, Lunch, and Meeting. To set a break reason, select it from the Agent Status dropdown in the agent panel.

Supervisors can see each agent’s current break reason in real time from the Agent Statistics :: Live view. Break reason data feeds into the Disposition Statistics report, helping supervisors track agent break patterns and total break time across shifts.

Predictive/Manual modes

ICTBroadcast provides options to agents to select the dialing method, either manual dialing (manual) or automated dialing (predictive). An agent can select the manual option to dial manually from a web phone or the select predictive option to let ICTBroadcast automatically dial the contact list

Show Dialogue/Scripts:

It provides you an option to display a campaign script/dialogue for each campaign.

Call on Hold

This is the option in ICTBroadcast for hold a call. An agent can hold during the live call to talk to other agents/persons to get updates/information for the customer.

Call Transfer to Another Agent

It provides an option to transfer calls, the agent can transfer the call to the desired extension. The agent can select the extension from the list by clicking the transfer button.

Attended Call transfer:

Attended call transfer is used when an agent put the call of customer on the hold and connects with another team member. The agent explains the case as well as the fact that he or she is transferring the call of the customer. Once he connects the customer to another team member and the conversation gets started, then only the main agent hangs up the call.

Blind call transfer:

Blind call transfer feature is used when an agent transfers the call to another team member without informing about the customer or case. In this case, it is also possible that the call
gets abandoned or the customer needs to share the same story again.

Multi-party Bridge

The Multi-party Bridge lets an agent conference a third party into an active call without ending the existing conversation. While on a call with a contact, the agent can dial a second number — such as a specialist, supervisor, or external resource — and bridge all three parties into a single call. This differs from a supervised transfer: the agent stays on the line throughout. Use Multi-party Bridge when you need to escalate or bring in expertise without abandoning the contact.

Campaign List:

ICTBroadcast provides an option to display a campaign list. An agent can monitor campaigns progress and status as shown

Contacts List:

ICTBroadcast provides an option to display the contact list. An agent can view contacts detail, also directly dial any person from the given list by click by the dial button. An agent can remove the desired contact from the list as shown:

Script:

ICTBroadcast provides an option to display script on screen. An agent can view script/content that being used in campaign as shown:

Update Contacts:

ICTBroadcast provides an option to update contact information. An agent can change/update contact information if required as shown:

Disposition:

ICTBroadcast provides an option to reschedule call times. In case of recipient do not take a call/busy, Agent can reschedule the call from the disposition option as shown:

The following reschedule time options are available:

10 Minutes: Retry the contact 10 minutes from now
30 Minutes: Retry the contact 30 minutes from now
1 Hour: Retry the contact 1 hour from now
Next Day: Schedule the retry for the following day
Next Week: Schedule the retry for the same time next week

Agents can also mark a call with a disposition outcome — Sale, DNC (Do Not Call), Busy, No Answer, or other campaign-defined results. Disposition data feeds directly into the Disposition Statistics report for supervisor review.

How to create a Dialogue script for agents in ICTBroadcast:

Admin/User can configure dialogue/personal scripts for agents in the following way:

Go to the Message menu from main-menu
Choose Dialogue/Script menu from sub-menu
Click the Add New button

Fill out all the fields and click Save.

Note: Users can make a dynamic/customized dialogue by using Tokens. Click on Show Available Tokens, to add tokens in dialogue.

How to assign a new Campaign to an agent

ICTBroadcast provides integration of Web-Phone in two campaigns only. To configure these campaigns you need to follow the steps given below:

Agent Campaign

Enter a Campaign Name
Select the contact group
Set outbound callerID
Set AMD option
Add URL of CRM for group

Both agent campaign and interactive campaigns can be configured to trigger a pop-up on the agent screen, the same URL should point to some CRM or another contact center. URL can be configured in the Contact URL field as shown in the following image

Select Dialogue

There is an option of Dialogue/Script under the general-options section in agent campaign or interactive campaign type where user can select recently created dialogues.

Webphone lets you customize the campaign script/ dialogue. It provides you an option to display campaign script/dialogue for each campaign. Agents can use this functionality by creating campaigns with dialogue.

Select Agent
Save the campaign

Interactive/Press 1 Campaign

Enter a Campaign Name
Select the contact group
Set outbound callerID
Set AMD option
Add URL of CRM for group

Both agent campaign and interactive campaigns can be configured to trigger a pop-up on the agent screen, the same URL should point to some CRM or another contact center. URL can be configured in the Contact URL field as shown in the following image

Select Dialogue

There is an option of Dialogue/Script under the general-options section in agent campaign or interactive campaign type where user can select recently created dialogues.

Webphone lets you customize the campaign script/ dialogue. It provides you an option to display campaign script/dialogue for each campaign. Agents can use this functionality by creating campaigns with dialogue.

Select Key for Transfer

On which key/button system should transfer the call to Agent / Extension

Select Agent
Save the campaign

Important Links :

The post ICTBroadcast Service Provider Edition Agent Guide (WebRTC) first appeared on ICTBroadcast.

ICTBroadast Version 4.1, Advance call center software for your needs

Tahir Almas — Thu, 07 Oct 2021 11:55:37 +0000

Even in today’s digital world, when reaching out to customer service, clients often prefer phone help. The immediacy and familiarity of talking to a real human being on the phone create trust between a client and a customer service agent. But, for many growing businesses, handling phone support can be a major challenge. Although it’s also a fantastic way to help customers, phone support is often the most effective way to help customers.
If you want to provide better phone customer support, you’ll need an efficient tech call center that supports your support service. Without apps, when agents fail to answer questions, clients will be left waiting on hold. Since they will not have any insight into the call frequency or patterns, managers will not be able to handle the confusion.
The best call center software routes call to suitable agents, offers employees the extra environment, and helps the administration perform an omnichannel strategy.

New And Advance Call Center Software in 2021

So, if you’re a team of ten or a few hundred, it’s important to choose the right resources to build your call center. This article breaks down the most popular call center software with its new and incredible features for all your needs in 2021. Yes! that’s ICTBroadcast with its latest release 4.1, all new features discuss below.
Advance WebRTC based agent panel introduced with call disposition support

Click to dial feature added in agent panel
IVR Studio transfer app enhanced with action on failed status
IVR Studio paly menu app enhanced with token/dynamic available support
Rest API of campaign report developed to fetch extra custom fields of different campaigns
Manual dialing, as well as Preview / Progressive dialing support, added
Automatic deletion of duplicate records from while uploading contacts as CSV file
Call back support added into integration
Push call status / Integration support added for invalid calls and tokens
Database cleaning script developed
Google TTS support added

Advance WebRTC based agent panel introduced with call disposition support

WebRTC based on HTML5 specification. It enables browsers to directly communicate with each other and send real-time audio, video, and data. Services such as Facebook Messenger, Slack, Houseparty, Webex, and now Skype are used to work from browsers. ICTBroadcast offers new and advanced WebPhone-based WebRTC features. This new ICTBroadcast WebPhone enables agents to accept calls through their browsers without any external hardware or software.

Click to dial feature added in agent panel

The agent can dial a contact number in the “Contact List” tab which he/she wants. There is an option “Dial” for each contact to dial manually.

IVR Studio transfer app enhanced with action on failed status

In IVR studio, the “Call Transfer” app improved by adding new action named “failed”. Using this option User/Admin can decide what to do next if the call transfer is failing.

IVR Studio paly menu app enhanced with token/dynamic available support

“Options Menu” app in IVR studio also improved the Token/Dynamic option, and also increase the total number of options e.g 1 to 0.

Rest API of campaign report developed to fetch extra custom fields of different campaigns

New rest APIs develop to generate campaign reports with extra custom fields of different campaigns.

Manual dialing, as well as Preview / Progressive dialing support, added

The agent can do manual dialing from the “Contact list” tab. There Agent can see all the contacts of the campaign and he/she can dial it manually as desire.

Automatic deletion of duplicate records from while uploading contacts as CSV file

When the User/Admin “Import” contacts using CSV, the latest ICTBroadcast released new feature will delete all duplicate contact(s) while saving new contacts and show the result that how many save and how many were deleted.

Call back support added into integration

Call back functionality added in Messages -> Integrations. Where admin can configure the call back setting. There is 2 type of call back functionality:

SOAP APIs
Custom HTTP request

Push call status / Integration support added for invalid calls and tokens

When something went wrong with the call or there is an invalid token, To handle this situation ICTBroadcast now supports Push call status / API Integration where the admin can select the desired API which he wants to call in these circumstances.

Database cleaning script developed

ICTBroadcast developers write the script that cleans up the database and removes unnecessary temp stuff to boost it. Admin/User can trigger this script from the System Configuration menu.

Google TTS support added

In the new ICTBroadcast release 4.1, ICTBroadcast now supports Google TTS (Text-to-Speech). Admin/User just needs to put GOOGLE TTS APIs access to use it.

The post ICTBroadast Version 4.1, Advance call center software for your needs first appeared on ICTBroadcast.

WebRTC Webphone for ICTBroadcast Auto Dialer

Tahir Almas — Thu, 07 Oct 2021 09:30:52 +0000

ICT Innovations brings another amazing feature for call centers and Agents’ support! WebPhone. ICTAgent supports Voice, SMS, Fax, and email communications technologies however here we will discuss only voice communications.

What is a WebPhone? and How does it help Agents?

Agents status (Ready/Busy)
Predictive/Manual
Call Status
End Call
Show Script
Call on Hold
Transfer Call to Selected Agent

Using Web-Phone

Users can even access integrated webRTC Web Phone by login in as an agent by giving an extension and password.

Set Agent Status

ICTBroadcast provides an option to change the status of Agent, the agent can set his status as ready and not ready by selecting an appropriate status option from the Agent status box,

Note: The agent with ready status can receive, forward, or make calls

ICTBroadcast provides an option to agents to select a dialing method, either manual dialing (manual) or automated dialing (predictive). An agent can select the manual option to dial manually from a webphone or the select predictive option to let ICTBroadcast automatically dial the contact list

Show Dialogue/Scripts

It provides you an option to display a campaign script/dialogue for each campaign.

Create Dialogue In ICTBroadcast

Agents can configure this functionality by creating campaigns with the dialogue option. You can create the dialogue/personal scripts in the following way:

Go to the Message menu from main-menu
Choose Dialogue/Script menu from sub-menu
Click the Add New button

Fill out all the fields and click Save.

Note: The user can make a dynamic/customized dialogue by using Tokens. Click on Show Available Tokens, to add tokens in dialogue.

How to Create a New Campaign with Web-Phone support

ICTBroadcast provides integration of WebPhone in two campaigns only.

Agent Campaign

Interactive/Press 1 Campaign

To configure these campaigns you need to follow the steps given below:

Add URL of CRM for group
Select a suitable Dialogue

Add URL of CRM for pop-up

Both agent campaigns and interactive campaigns can be configured to trigger a pop-up on the agent screen, the same URL should point to some CRM or other contact centers. URL can be configured in the Contact URL field as shown in the following image

Select Dialogue/Script

There is an option of Dialogue/Script under the general-options section in the agent campaign or interactive campaign type where users can select recently created dialogues.

Webphone lets you customize the campaign script/ dialogue. It provides you an option to display the campaign script/dialogue for each campaign. Agents can use this functionality by creating campaigns with dialogue.

CREATE AGENT CAMPAIGN

ICTBroadcast provides integration of WebPhone in 2 Campaigns. It can be used with Agent and Interactive Campaigns.

To configure these campaigns you need to follow the steps below:

CONTACT / CRM URL FOR POPUP

You can utilize the webphone features in Interactive Campaigns and Agent Campaigns.

When using WebPhone, Agent and interactive campaigns can be configured to trigger a Popup on the agent screen, the same URL should point to some CRM or other contact provider app. URL can be configured in the Contact URL field as shown in the following image.
DIALOGUE

Creating your own Dialogue/Script can be done in a few steps now

Dialogues are specific to agents and interactive campaigns, in these campaigns an option of Dialogue / Script is given under the General options section and the user can select recently created dialogue here.

USING WEBPHONE

Users can even access integrated webphones by selecting the desired extension from the top Agent Panel dropdown.

SET AGENT STATUS

ICTBroadcast provides an option to change the status of the Agent. You can set the status ready or not ready

Note: Agents with ready status can receive / makes/transfer calls

CALL TRANSFER

Transferring a call is not an issue now as the call can be transferred to the desired extension by clicking the call transfer button and then selecting the desired extension

LOAD CONTACT

You can view and edit prospect details by clicking on load contact.
Please note that it is different from the popup URL. Instead of a popup, it will redirect the current window to ICTBroadcast’s internal contact record where the user can edit/update it accordingly.

KEYPAD / DTMF

Handy keys are provided to the agent for operating the webphone.
A-List of usable key sequences that agent can use for different purposes is given below: For hangup press *0

For call transfer, press #1 and then enter the target extension number alternatively (without pressing any key) agent can select the target number from the transfer dropdown button.

For attended call transfer press #2 followed by the target extension

The post WebRTC Webphone for ICTBroadcast Auto Dialer first appeared on ICTBroadcast.

How to use WebRTC Phone for Agents in Interactive / Press 1 campaign

Tahir Almas — Thu, 07 Oct 2021 06:18:42 +0000

A WebPhone is a WebRTC-based phone that allows agents to accept incoming and make outbound calls using their browsers without any external hardware or software. ICTBroadcast provides integrated WebPhone for Agents. ICTBroadcast has a WebPhone with incredible features for call centers and Agent support! ICTAgent supports Voice, SMS, Fax, and email communications.

How to use WebRTC Phone in ICTBroadcast

ICTBroadcast presents its new WebPhone based on WebRTC with new features and a user-friendly UI for agents. Agents just need to go to their browser and log in as agents. Here you are watching the most feature-rich WebPhone. It allows agents to perform the following actions:

Log in as Agent
Agent status (Ready/Busy)
Predictive/Manual modes
Show Script
Call on Hold
Transfer Call to Another Agent
Campaign List
Contact List
Script
Update Contact
Disposition/Re-Schedule

Create Dialogue/Scripts for Agent

Admin can provide dialogue/script to agents to display for each prospect. Admin/User can set script functionality while creating campaigns by using Dialogue / Script options. First, they need to create the script.

How to Create Dialogue / Script in ICTBroadcast

In ICTBroadcast, click on Dialogue / Script under the Message menu. Then click on Create new.

In the Dialogue / Scripts form, Dialogue name and Dialogue / Script fields are required, though Description is an optional field. Fill out the form’s fields and click Save. Note: Users can make a dynamic dialogue by using Tokens. To add tokens in dialogue, please click on Show available tokens.

How to Create New Campaign with Web-phone support

ICTBroadcast supports two integrated campaigns with WebPhone:

Agent Campaign
Interactive / Press 1 Campaign

To set up the above-mentioned campaigns, follow these steps:

Add URL of CRM for a popup
Select a suitable Dialogue

Add URL of CRM for a popup

When using WebPhone, Agent campaign and interactive campaigns can be configured to trigger a popup on the agent screen. The same URL should point to some CRM or other contact center. URL can be configured in the Contact URL field as shown in the following image. Users can also use contact tokens to build contact URL.

Select Dialogue / Script

In Agent Campaign / Interactive Campaigns, there is an option of Dialogue / Script under the General options section. Users can select the pre-created dialogue here.

Set Agent Status in WebRTC WebPhone

ICTBroadcast provides an option to change the status of an Agent. The agent can set their status as ready or not ready by selecting the appropriate status option from the Agent status box.

Note: An agent with ready status can receive, forward, or make calls.

Call Transfer in new WebRTC WebPhone

There is an option of Dialogue/Script under the general-options section in agent campaigns or interactive campaigns where users can select recently created dialogues.

Update Contacts

ICTBroadcast provides the option to update contact information. The agent can change/update contact information if required as shown below:

DTMF/NumPad

A useful numpad is provided along with the new WebPhone. The following is a list of usable main sequences that the agent will use for various purposes, e.g., 0 for Hangup. Press #1 for a call transfer.

Forward to Agent on Answer

Creating Personalized messages using custom tokens in SMS campaign

Steps to run a Press 1 campaign / interactive voice broadcasting

Note: Please configure/set up gateways/SIP providers, add contacts and voice recordings, and then follow these instructions. In the case of ICTBroadcast Service Provider Edition, please set up billing too.

Add and configure agents

Visit https://your-domain.com/login.php?action=user
Login as admin
Create an extension in ICTBroadcast

Go to Administration menu -> Click on Agent / Extensions -> Click on New Extension button. Enter Agent Name, Agent Number, and Password, then click on Save button.

Create Queue / Skill Group

Go to the Administration menu -> Click on Agent / Extensions -> Click on New Extension button. Enter Agent Name, Agent Number, select Skill Group from Forward To, and select Agents for this Queue / Skill Group. Then click on the Save button.

Assign extension/Queue to user

Go to the Administration menu -> User Management -> Click on Assign Extensions of any user under the OPERATIONS column. Select agents and skill groups and then click on the Assign button.

Login as Agent to Receive Calls in ICTBroadcast

If you want to log in as an Agent, use a different computer/browser and perform the following steps:

Visit https://your-domain.com/login.php
Click on Agent Login
Enter agent number and agent password
Click on the ready button from the agent phone portal
When the agent is ready, you can listen to music in the background

How to Start Interactive Campaign

Open the Navigation menu and select the main menu “Campaigns”.
Select the submenu “Campaign Management” under the main menu and click on it.
Click on the Add new button on the top right side. A page will open. Select the Interactive Campaign from it.

Enter the required information in the desired field, such as the campaign name.
Select the contact group and caller ID.

Select the Agent to which the campaign will be assigned and the Agent campaign wrap-up timeout.