|Subject:||Re: [Linphone-developers] JsSIP and Linphone issue|
|Date:||Fri, 26 Jun 2020 11:59:10 +0200|
|User-agent:||Mozilla/5.0 (X11; Linux x86_64; rv:68.0) Gecko/20100101 Thunderbird/68.6.0|
I do not know precisely how to talk to WebRTC, and the details about SAVPF.
I will try to gather some information from my colleagues.
As of now, I can give you general advice, based on my intuition.
We have a wiki
page describing how to configure Linphone to work well with
Did you try with DTLS, (mabe also SIP/TLS) and ICE enabled in Linphone? (DTLS settings can be enabed in settings > call and ICE can be enabled in settings > network)
Junior Software Engineer
Belledonne Communications, the company behind Linphone
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.
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