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From: | Peio Rigaux |
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 |
Hello.
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
WebRTC.
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)
Regards,
Peio Rigaux
Junior Software Engineer
Belledonne Communications, the company behind Linphone
Linphone.org
Hi,
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.
Thanks,
Meghana J
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