Baresip with WSS

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Altyn Altynov

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Jan 9, 2024, 5:31:00 AM1/9/24
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I'm trying to use WSS via baresip(on android) with FreeSwitch.
WSS via jssip clients with the same FreeSwitch works well.
In logs on both sides I see that rtp and dtls are sent, but there is no audio and video shown, and I also can't see any sent rtp packets in wireshark. 
We can't find any problems in the logs; any idea where we can look? 

Here is the registration code:
val uri =
            "<sip:${account.login}@${account.sipHost}:${sipPort};transport=wss>;" +
                    "stunserver=\"stun:${account.stunHost}:${account.stunPort}\";" +
                    "mwi=no;mediaaf=ipv4"
        var userId = BareSipNative.ua_alloc(uri)
        val accountId = BareSipNative.ua_account(userId)
        BareSipNative.account_set_auth_user(accountId, account.login)
        BareSipNative.account_set_display_name(accountId, account.login)
        BareSipNative.account_set_auth_pass(accountId, account.password)
        BareSipNative.account_set_regint(accountId, 3600)
        BareSipNative.account_set_audio_codecs(accountId, "PCMA/8000/1")
        BareSipNative.account_set_video_codecs(accountId, "H264,H265,VP8,VP9,AV1")
        BareSipNative.ua_update_account(userId)
        BareSipNative.ua_register(userId)


And here are the logs from creating the baresip instance to accepting the incoming call
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