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Re: [Linphone-developers] JsSIP and Linphone issue


From: German Cancio
Subject: Re: [Linphone-developers] JsSIP and Linphone issue
Date: Fri, 26 Jun 2020 14:14:53 +0000


As a quick comment, having this kind of integration working is not trivial and is certainly not just a one-place config issue; it may require changes on both end points. There are other parameters that you may want to check out, such as RTCP multiplexing.

Are you connecting directly to your web app UA - or via a proxy such as Kamailio and/or a B2BUA like Asterisk?

BR Germán


On 26 Jun 2020, at 15:39, Meghana Jagtap <meghana.jagtap@springct.com> wrote:

Hi,

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.

Thanks,

Meghana J


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

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

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

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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