<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><blockquote type="cite"><div>From that I would like to use iLBC codec to start a call between users. (iLBC seems to be the most supported format)<br></div></blockquote><div><br></div>Note that the "LIVE555 Streaming Media" software does not include any codecs (encoding or decoding software). So you would need to supply your own iLBC codec.</div><div><br></div><div>We do, however, support the RTP Payload Format for iLBC audio (as defined in RFC 3952), using the "SimpleRTPSink" class (for transmission) and the "SimpleRTPSource" class (for reception).</div><div><br></div><div><br></div><div><blockquote type="cite"><div>- Can we initiate a rtp/rtcp protocol with an already opened udp socket ?<br></div></blockquote><div><br></div><div>No. However, if you have a port *number*, you can create a "Groupsock" object to use that port number. Its constructor will create the UDP socket for the given port number.</div><div><br></div><div>Once you have a "Groupsock" object, you can use this to create a "SimpleRTPSink" object (for RTP transmission), or a "SimpleRTPSource" object (for RTP reception). You can also use a separate "Groupsock" object - at each end - to create a "RTCPInstance" object (to implement the RTCP protocol). Note that by convention, RTP uses an even-numbered socket, and RTCP uses the next socket (i.e., an odd-numbered socket). This is not something that's universally enforced, but it's something that you should keep in mind, and try to follow if you can.</div></div><div><br></div><div><br></div><div><blockquote type="cite"><div>- In order to make a voip app, is it necessary to have a rtp server and rtp client on both devices (client 1 connecting to server 2 / client 2 connecting to server 1)?<br></div></blockquote><div><br></div>You would need both a "SimpleRTPSink" object and a "SimpleRTPSource" object on each device.</div><div><br></div><div><br><blockquote type="cite"><div>- Using live555, seems to me like using a bazooka to kill a fly. Live555 is so much complete that it seems huge comparing to what I need to do, isn't it ?<br></div></blockquote><div><br></div>Yes. Much of the code deals with the RTSP protocol, which is not relevant for your application. Note also that we do not implement SIP, which is the most commonly used control protocol for VoIP. (We do have a rudimentary SIP client implementation, but it falls far short of what you'd need for VoIP.) So, if you're planning to use SIP, you should definitely look elsewhere.</div><br><br><div apple-content-edited="true">
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; font-size: medium; ">Ross Finlayson<br>Live Networks, Inc.<br><a href="http://www.live555.com/">http://www.live555.com/</a></span></span>
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