<div dir="ltr">Update: We integrated pjsip with WebRTC on iOS, we are using pjsip for the signalling part and passing the SDP back and forth to and from WebRTC. <div><br></div><div>Our current issue is that after initializing the call we can't hear any audio (should hear moh at this point), everything looks OK on Freeswitch until we start seeing alot of those errors:<div><div>2015-01-16 20:25:31.111260 [ERR] mod_opus.c:418 Decoder Error: corrupted stream fs:4080 plc:0!</div><div>2015-01-16 20:25:31.111260 [ERR] switch_core_io.c:697 Codec RAW Signed Linear (16 bit) decoder error! [9]</div><div><br></div><div>Any idea what might be wrong here?</div><div><br></div><div>Thanks,</div><div>Adam</div><div><br></div></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On 15 January 2015 at 15:27, Adam Ben-Ayoun <span dir="ltr"><<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank">adam.ben.ayoun1@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">In our case, all clients are calling to the conference on FreeSwitch, hence, no registration is required at the moment, but it definitely opens up other options for future.</div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On 10 January 2015 at 20:38, Adam Ben-Ayoun <span dir="ltr"><<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank">adam.ben.ayoun1@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Great to hear that. Thanks again.</div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On 10 January 2015 at 20:35, Carlos Ruiz Díaz <span dir="ltr"><<a href="mailto:carlos.ruizdiaz@gmail.com" target="_blank">carlos.ruizdiaz@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">WebRTC doesn't specify a signalling protocol. This means that you can use SIP over any transport you want to carry the webRTC enabled SDP. </p>
<p dir="ltr">FS will receive the SDP, detect that has a RTP/SAVPF profile and start handling it accordingly.</p>
<p dir="ltr">Take for example Jitsi or IMSDroid, they both support webRTC and do SIP over UDP/TCP/TLS. </p>
<p dir="ltr">Regards, <br>
Carlos </p><div><div>
<div class="gmail_quote">On Jan 10, 2015 12:08 PM, "Adam Ben-Ayoun" <<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank">adam.ben.ayoun1@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Thanks Anthony. I assume that means I can use SIP over TCP/TLS for signalling? Also, will mandatory WebRTC requirements such as DTLS-SRTP work when communicating with FS (when stuff like fingerprint, etc)?</div><div class="gmail_extra"><br><div class="gmail_quote">On 10 January 2015 at 19:49, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">The WebRTC media engine is driven completely by the SDP, the transport will not make any difference.<div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Fri, Jan 9, 2015 at 5:26 PM, Adam Ben-Ayoun <span dir="ltr"><<a href="mailto:adam.ben.ayoun1@gmail.com" target="_blank">adam.ben.ayoun1@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">Hi,<div><br></div><div><div>We are developing a mobile client that will use the WebRTC media stack and Freeswitch as an MCU (only for conference calls). My question is, since we build a native app, can we use SIP over TCP for signalling? In other words, if Freeswitch receives the WebRTC kind of SDP, will it be able to communicate in the same way as if we were using the SIP over Websocket (the other Freeswitch option)? Any corner cases/considerations with this? Our goal is to avoid implementing SIP over Websocket on the client as much as possible.</div></div><div><br></div><div>Thanks,</div><div>Adam</div></div>
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