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It sounds like your ACK is possible not matching the transaction and
never routing, then the stream is shutdown based on timeout.<br>
<br>
Check your signaling to see if it looks like your ACK/200 are being
retransmitted.<br>
<br>
rtpproxy is ... pretty "dumb" it ... all it does is forward
packets. It doesnt care if they are encrypted or not.<br>
<br>
-Eric<br>
<br>
<div class="moz-cite-prefix">On 02/13/2015 11:27 AM, Semen Golubcov
wrote:<br>
</div>
<blockquote
cite="mid:CAKTAW0sW=OGGqaHWC124svy+JhmZWDfZW5pPYDz51RHoncM11A@mail.gmail.com"
type="cite">
<div dir="ltr">Hello! I'm using the latest opensips with latest
rtpproxy :<br>
<br>
Basic version: 20040107<br>
Extension 20050322: Support for multiple RTP streams and MOH<br>
Extension 20060704: Support for extra parameter in the V command<br>
Extension 20071116: Support for RTP re-packetization<br>
Extension 20071218: Support for forking (copying) RTP stream<br>
Extension 20080403: Support for RTP statistics querying<br>
Extension 20081102: Support for setting codecs in the
update/lookup command<br>
Extension 20081224: Support for session timeout notifications<br>
Extension 20090810: Support for automatic bridging<br>
<br>
i did setup the secure tls connection between clients in the
opensips, it uses the client certificate verification. But the
interaction with rtpproxy is getting messed up somehow. I'm
using the blink softphone to test on. So i have 2 accounts: bob
and alex. When i do <b>call</b> <b>bob</b> <b>from</b> <b>alex</b>
i get this kind of behaviour in the <b>bob's softphone</b>:<br>
<img src="cid:part1.09020202.08090202@uphreak.com" height="37"
width="404"><br>
So it hangs on "starting media", but in the same time the
actual connection is established, me and my partner can hear
each other and we can talk perfectly, so i assume the actual
stream is allright. It hangs for exactly 15 seconds then we get:<br>
<img src="cid:part2.06070903.00000800@uphreak.com" height="23"
width="404"><br>
<br>
<p style="text-align:left;clear:both"><b>"media stream timed out
while starting" in the bob's softphone</b>.</p>
<p style="text-align:left;clear:both"> <b>On the alex softphone
side</b>:</p>
<p style="text-align:left;clear:both"><img
src="cid:part3.00050304.01080803@uphreak.com" height="26"
width="404"><br>
<br>
</p>
<p style="text-align:left;clear:both"> "call ended by remote". <br>
</p>
<p style="text-align:left;clear:both">The icon indicating that
rtp stream is encrypted is not shown on bob's side, but the
stream is working. I tried to disable tls, and use plain tcp,
and it's working fine without tls, call is not getting stuck
and automatically terminate. <br>
</p>
<p style="text-align:left;clear:both">My rtpproxy is running
like so:</p>
<p style="text-align:left;clear:both">rtpproxy -u rtpproxy -s
udp:localhost:12221</p>
<p style="text-align:left;clear:both">And my opensips config is
generated by osipconfig utility (i didn't modify the routes at
all) see attached opensips.cfg<br>
</p>
<p style="text-align:left;clear:both">I looked up the syslog for
rtpproxy entries, apparently there is no entries from rtpproxy
or opensips with any error, other than</p>
<p style="text-align:left;clear:both">Feb 13 04:00:24 user
/usr/local/sbin/opensips[17535]:
ERROR:rtpproxy:force_rtp_proxy: Unable to parse body</p>
<p style="text-align:left;clear:both">Feb 13 04:00:22 user
/usr/local/sbin/opensips[17535]: DBG:tm:matching_3261: RFC3261
transaction matching failed<br>
</p>
I hope someone can help me to solve this problem, i'am hanging
on this for a week. If necessary i can post the syslog and
blink softphone sip traces, of our test conversation. Maybe the
problem is with ACK? <br>
</div>
<br>
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<br>
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</pre>
</blockquote>
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