<div dir="ltr"><div style="font-family:arial,sans-serif;font-size:13px">Hi again,</div><div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">Can how can insert from opensips, without any other pbx, b2b ..., 183 Progress with sdp. The ideea is, that sometime i need to insert in media stream at begining, some audio, using rtpproxy and for this i should have sdp and rtpproxy already used in call. As application, for example i want to change the ringback tone with opensips and rtpproxy from clasic ringing to some music or message. </div>
<div style="font-family:arial,sans-serif;font-size:13px">Thanks in advance for those who will tell me that opensips is sip proxy and has nothing to do with relayed media. </div><div style="font-family:arial,sans-serif;font-size:13px">
</div><div style="font-family:arial,sans-serif;font-size:13px">the call trace case is next one:</div><div style="font-family:arial,sans-serif;font-size:13px"><br></div><span style="font-family:arial,sans-serif;font-size:13px">(A)invite ->opensips ->invite(B)</span><br clear="all" style="font-family:arial,sans-serif;font-size:13px">
<div style="font-family:arial,sans-serif;font-size:13px">(A)trying <-opensips <-trying(B)</div><div style="font-family:arial,sans-serif;font-size:13px">(A)ringing <-opensips <-ringing(B)</div><div style="font-family:arial,sans-serif;font-size:13px">
(A)progress <-opensips</div><div style="font-family:arial,sans-serif;font-size:13px">(A)200ok <-opensips <-200OK(B)</div><div style="font-family:arial,sans-serif;font-size:13px">(A) ACK ->opensips ->ACK(B)</div>
<div><br></div>-- <br>Dani Popa
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