Stranger. Try hook SRTP in Wireshark.
I am getting same without "add to registrar uri and call peer uri ;transport=tls;lr «
Simple try
"Try add to registrar uri and call peer uri ;transport=tls;lr "
I already have this in my call uri and register uri. Call is established but video and audio not working. Without this setting I am getting call without voice too. Try add to registrar uri and call peer uri ;transport=tls;lr Me help that, but ONLY in release. In debug no voice over TLS
I am able to make the call. Just the video and audio is not working. TLS setting is correct only as you can see in the logs, it's communicating via TLS only. and those certificates are optional I guess as Linphone is working fine without those certificates. When creating acc with TLS I am setting
val tlsCfg = TlsConfig() tlsCfg.certFile = certPath tlsCfg.privKeyFile = certPath tlsCfg.verifyServer = false tlsCfg.method = pjsip_ssl_method.PJSIP_TLSV1_METHOD accCfg.mediaConfig.srtpUse = pjmedia_srtp_use.PJMEDIA_SRTP_MANDATORY accCfg.mediaConfig.srtpSecureSignaling = 1 accCfg.mediaConfig.transportConfig.tlsConfig = tlsCfg
Adding ;transport=tls;lr to registrar uri and uri when making call Hope this help you
More on that is using two instances of linphone I am able to make the video call fine. If i turn of SRTP and use RTP in PJSIP everything works fine. Only when using SRTP it's creating the problem. The way I am configuring is like below:
ua_cfg.use_srtp = PJMEDIA_SRTP_MANDATORY; ua_cfg.srtp_secure_signaling = PJSUA_DEFAULT_SRTP_SECURE_SIGNALING;
pjsua_srtp_opt srtp_opt; pjsua_srtp_opt_default(&srtp_opt);
ua_cfg.srtp_opt = srtp_opt; ua_cfg.srtp_optional_dup_offer = PJ_TRUE;
It looks like it also does not work in android. This is the exact problem I am facing in ios. Please help regarding this. What can be the isssue? I am able to register and get the call. How can I get that if the settings are not correct. Only video and audio is not working.
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