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[Linphone-developers] JsSIP and Linphone issue

From: Meghana Jagtap
Subject: [Linphone-developers] JsSIP and Linphone issue
Date: Fri, 26 Jun 2020 12:46:30 +0530
User-agent: Mozilla/5.0 (X11; Linux x86_64; rv:60.0) Gecko/20100101 Thunderbird/60.9.0


I am trying to make a call from Linphone Desktop app to my web app that is build upon JsSIP. The call gets connected rarely and fails most of the time. The frequency of failure is 8 out of 10. SIP Error code - ' 488 Not Acceptable here' can be seen in the Linphone logs.

Upon searching the logs it was found that Linphone's SDP contains RTP/AVPF in audio m lines, which is not compatible with the WebRTC standards. WebRTC requires SRTP(Secure RTP), ICE, a new SDP profile (SAVPF).

Is this the reason for failure. Please help me in resolving this issue.

Meghana J

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