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SIP (Session Initiation Protocol) is an important industry standard protocol that is widely used for establishing, modifying and terminating multimedia sessions or voice and video calls across IP networks. It allows for a high degree of flexibility regarding codecs, and media types that can be interchanged during the session. A SIP Engineer is a skilled software developer/programmer that specializes in network collaboration protocols like SIP, RTSP and RTP that help enable peer-to-peer communication over the internet, reducing costs and allowing for more efficient deployment of voice and video conferencing.
A SIP engineer can do many things ranging from building Voice over IP systems to creating specialized multimedia applications. They will build sophisticated systems with various components such as audio encoders/decoders, media gateways, signaling gateways and user agent clients. They are also responsible for configuring and optimizing the system to achieve the best results and creating real-time simulations like teleconferencing or IVRs (interactive voice response).
Here’s some projects that our expert SIP Engineer made real:
- Secure upgrades for open source communications protocols
- Debugging and solving complex VoIP issues
- Carrier grade server implementations in cloud environments
- System wide port configuration optimization
- Integrating complex third party applications with existing softwares
- Coordinating German DIDs/Voip Numbers with PBXs
- Developing automated communications using Python
SIP Engineers are always in demand as great knowledge in this protocol is essential for setting up reliable communication services with low latency at an affordable rate. At Freelancer.com you can hire an experienced SIP Engineer who not only understands the inner workings of current standards but also able to keep up with new developments in the communication world. Post your project today and get the expertise you need to create a powerful yet cost effective communications environment!
Dari 549 ulasan, klien menilai SIP Engineers kami 4.9 dari 5 bintang.Rekrut SIP Engineers
SIP (Session Initiation Protocol) is an important industry standard protocol that is widely used for establishing, modifying and terminating multimedia sessions or voice and video calls across IP networks. It allows for a high degree of flexibility regarding codecs, and media types that can be interchanged during the session. A SIP Engineer is a skilled software developer/programmer that specializes in network collaboration protocols like SIP, RTSP and RTP that help enable peer-to-peer communication over the internet, reducing costs and allowing for more efficient deployment of voice and video conferencing.
A SIP engineer can do many things ranging from building Voice over IP systems to creating specialized multimedia applications. They will build sophisticated systems with various components such as audio encoders/decoders, media gateways, signaling gateways and user agent clients. They are also responsible for configuring and optimizing the system to achieve the best results and creating real-time simulations like teleconferencing or IVRs (interactive voice response).
Here’s some projects that our expert SIP Engineer made real:
- Secure upgrades for open source communications protocols
- Debugging and solving complex VoIP issues
- Carrier grade server implementations in cloud environments
- System wide port configuration optimization
- Integrating complex third party applications with existing softwares
- Coordinating German DIDs/Voip Numbers with PBXs
- Developing automated communications using Python
SIP Engineers are always in demand as great knowledge in this protocol is essential for setting up reliable communication services with low latency at an affordable rate. At Freelancer.com you can hire an experienced SIP Engineer who not only understands the inner workings of current standards but also able to keep up with new developments in the communication world. Post your project today and get the expertise you need to create a powerful yet cost effective communications environment!
Dari 549 ulasan, klien menilai SIP Engineers kami 4.9 dari 5 bintang.Rekrut SIP Engineers
Twilio/Asterisk Call Flow Troubleshooting
6 Hari left
Hi, My call flow is as follows: The call starts from the Twilio SIP trunk. Under this SIP trunk, I have configured my Asterisk server. Every call begins when the user dials the Twilio number, and it is received by the Asterisk server. My Asterisk server immediately calls the Twilio SIP domain. Under the Twilio SIP domain, I have a webhook that routes the call to my worker. If my Twilio client’s workers do not respond within 30 seconds, I use Twilio Refer to send the call back to my Asterisk server, which then connects to my backup SIP. The backup SIP is a simple SIP for Linphone. Flow: Caller → Twilio SIP Trunk → Asterisk Server → Twilio SIP Domain → Worker (if available) If no response from workers: Twilio SIP Domain → Asterisk Server → Backup SIP (L...
$592
Average bid
Twilio SIP Call Optimization Specialist
3 Hari left
Hi, Please understand the problem we are facing with a specific use case. You need to have knowledge of Twilio, SIP, and Asterisk. We are trying to achieve the following use case: A caller dials a Twilio number. After some time or under certain conditions, Twilio forwards the call to an Asterisk server. Asterisk receives the call from Twilio and immediately connects the caller to sip:test (essentially bridging the caller directly to the SIP address). Once the call is connected between the caller and sip:linphone, Twilio should no longer be involved in the call (to avoid duration charges). The desired flow is: Caller → Twilio → Asterisk → sip linphone At the end, the connection should be: Caller → sip inphone We understand that we will need to pay Twilio for the i...
$597
Average bid
Multi-Tenant VoIP System Development
3 Hari left
Terverifikasi
VoIP System Development Task Document Project Overview Build a multi-tenant VoIP system that allows users to add and manage their own SIP trunks, configure custom Caller IDs and prefixes, and track minute usage. The system must provide billing for users and reconcile minutes with providers. This will be a scalable, secure, and feature-rich platform capable of handling thousands of users and trunks. --- Project Goals 1. User Management: Support 1,000+ users, each managing up to 50 trunks. 2. Custom Configurations: Enable users to set custom prefixes and Caller IDs for each trunk. 3. Billing System: Track and bill users for minutes used and reconcile minutes with the provider. 4. Monitoring: Provide real-time metrics for active calls and system performance. 5. Scalability: Hand...
$256
Average bid
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