[Top][All Lists]

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

Re: [Linphone-developers] JsSIP and Linphone issue

From: Meghana Jagtap
Subject: Re: [Linphone-developers] JsSIP and Linphone issue
Date: Fri, 26 Jun 2020 19:09:52 +0530
User-agent: Mozilla/5.0 (X11; Linux x86_64; rv:60.0) Gecko/20100101 Thunderbird/60.9.0


As per your suggestion I tried DTLS and ICE enabled.

Enabling the DTLS, stopped the Linphone app functioning whereas enabling ICE slightly reduced the number of failure attempts. But more or less, I am looking for a one place solution, so that all the users need not make changes to their app.


Meghana J

On 26/06/20 3:29 PM, Peio Rigaux wrote:


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)


Peio Rigaux
Junior Software Engineer
Belledonne Communications, the company behind Linphone

Le 26/06/2020 à 09:16, Meghana Jagtap a écrit :


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

Linphone-developers mailing list

Linphone-developers mailing list

reply via email to

[Prev in Thread] Current Thread [Next in Thread]