Not working after adding also. TLS does not have any problem I guess. It's the SRTP which has the problem.
On Thu, Jul 18, 2019 at 7:38 PM Александр Клейменов <a.kleymenov@xxxxxxxx> wrote:
I am getting same without "add to registrar uri and call peer uri ;transport=tls;lr «_______________________________________________Simple try18 июля 2019 г., в 17:05, Anuran Barman <anuranbarman@xxxxxxxxx> написал(а):"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._______________________________________________On Thu, Jul 18, 2019 at 7:33 PM Александр Клейменов <a.kleymenov@xxxxxxxx> wrote:Without this setting I am getting call without voice too._______________________________________________Try add to registrar uri and call peer uri ;transport=tls;lrMe help that, but ONLY in release. In debug no voice over TLS18 июля 2019 г., в 16:58, Anuran Barman <anuranbarman@xxxxxxxxx> написал(а):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._______________________________________________On Thu, Jul 18, 2019 at 7:25 PM Александр Клейменов <a.kleymenov@xxxxxxxx> wrote:When creating acc with TLS I am setting_______________________________________________val tlsCfg = TlsConfig()tlsCfg.certFile = certPathtlsCfg.privKeyFile = certPathtlsCfg.verifyServer = falsetlsCfg.method = pjsip_ssl_method.PJSIP_TLSV1_METHODaccCfg.mediaConfig.srtpUse = pjmedia_srtp_use.PJMEDIA_SRTP_MANDATORYaccCfg.mediaConfig.srtpSecureSignaling = 1accCfg.mediaConfig.transportConfig.tlsConfig = tlsCfgAdding ;transport=tls;lr to registrar uri and uri when making callHope this help you18 июля 2019 г., в 16:42, Anuran Barman <anuranbarman@xxxxxxxxx> написал(а):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?Android Similar Problem:https://stackoverflow.com/questions/56031734/how-to-enable-srtp-with-pjsip-in-android_______________________________________________On Thu, Jul 18, 2019 at 5:26 PM Anuran Barman <anuranbarman@xxxxxxxxx> wrote: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.On Thu, Jul 18, 2019 at 5:24 PM Александр Клейменов <a.kleymenov@xxxxxxxx> wrote:Are you sure in account settings for TLS?_______________________________________________18 июля 2019 г., в 14:27, Anuran Barman <anuranbarman@xxxxxxxxx> написал(а):Hi, No even in release it is not working. Same problem._______________________________________________On Thu, Jul 18, 2019 at 4:53 PM Александр Клейменов <a.kleymenov@xxxxxxxx> wrote:Hello!
A have same problem on Android in debug, in release work nice - try release.
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