Hello Everyone, I need somebody who can compile again PJSIP library for PJSUA with all support library similar to following link with latest OpenSSL library. CSipSimple Sample: [url removed, login to view] Work expected: - Make necessary changes to successfully compile the PJSIP source code that is available in the above link. - Update
Based on open source Pjsip to build a softphone. To enable Audio layer on Rpi, and use USB mic. No desktop GUI, only a service daemon, and simple web interface or API to interact. Video support as well. Can consider based on Linphone as well Please check attached PDF file here. There are a lot of tutorials and instructions how to enable audio
We have several SNOM M25 and M65 Phones connected to a SNOM M700 Base. The Snom uses FreePBX Asterix 14.0.x (latest) as SIP-Gateway via PJSIP connected to SIP Gateway of German Telekom. All the Phones are properly connected and the Peers registered. Somehow, we can't manage to dial out or to be dialed in and we don't know why. Need an SIP Expert to
Hi Rajesh T., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I am looking for xmpp and pjsip experts for my projects. Reply me or Sen message to [url removed, login to view]@[url removed, login to view]
I'm looking for Android and iOS developers to make a social chat/calls/videocalls/paycalls app. We already have the design, so you have to program everything. IMPORTANT: The connection for the calls, pay calls (to landlines and mobiles) and video calls must be using PJSIP on our own servers, we will give you the access to them to makes test
We've the exact same problem as this question on stackoverflow [url removed, login to view] If we answered the video call from CallKit, we'll get OpenGL renderer initialization error and not be able to enable video after switch from CallKit to the app. Here is the log: 11:38:56.632 pjsua_call
We are trying to get customization to PJSIP source code to be able to do transferring using PJSIP within Asterisk. I have a requirements doc but need to interview anyone interested in the project before I send it to them.
Need an expert in PJSIP Library to complete a video rendering project. We've gotten almost 3/4 of the way but are missing some key components. The ideal candidate for this project is well versed in rendering video on mobile for a two way/multiplex call.
Expert in C, C++, PJSIP Stack Must be able to analyze the Wireshark captures to identify any SIP signaling/media issues. Should have worked on Server side Should be aware of RFC 3261,3264, NAT traversal, Media Codecs AMR, G729, Opus.
...CSIPSIMPLE using pjsip. We need to integrate vpn client into this. 2. Tunnelling Server - Linux based a. Bandwidth optimization b. Secure connection and communication c. SNMP monitoring Please bid only if you have developed something similar and tested and to provide us details and the reference of your developed required: Android, iOS Please
Modify the code so it works with PJSIP and Attended transfer within the asterisk C code. We can provide more details for the right developer.
Need Application Developer who has worked on PJSIP for mobile app for IoS and Android Simple app with standard screens - 1. Theam change 2. Login change 3. Voice mail 4. Authentication pattern change 5. Contact 6. Setting.
Hi Freelancers. I need to complete Chat/Voice/Video call app now. Of course I have source code (Android/iOS/Backend). Backend is python. I will share all source code to the right candidate. Please write your favorite fruit at the top of your cover letter so that I know you are not robot. Only bid who has 100+ hours of work history. Thanks