<div dir="ltr"><div><div><div>Babil,<br><br></div>You are right. <br><br></div>Semen,<br><br></div>You could probably attach the signalling to both side from opensips server, this will give more idea what exactly is happening. This something related to signalling only.<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Feb 14, 2015 at 8:08 AM, Babil <span dir="ltr"><<a href="mailto:gsbabil@gmail.com" target="_blank">gsbabil@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">Hi Semen,<br>
what is your network configuration? If the RTP-Proxy machine is an<br>
Amazon EC-2 (or alike) instance where an "elastic" public IP is attached<br>
to the Ethernet interface which has a different internal DHCP provided<br>
address (or you have a port-forwarding scenario), you'd need to use a<br>
patched version of RTP-Proxy.<br>
<br>
The RTP-Proxy patch, as shown here [0], will allow you to have an<br>
additional '-A' option to modify the advertised IP address (external<br>
publicly visible IP address) by RTP-Proxy which should resolve the<br>
timeout issues you are experiencing. This version [1] on Github also<br>
seems to have the patch integrated. You'll find a longer discussion on<br>
the RTP-Proxy and NAT problem here [2].<br>
<br>
Hope this helps. Good luck.<br>
<br>
P.S.<br>
In order to check your network address dissimilarities, you may try the<br>
following commands. I'm considering your external interface is `eth0`:<br>
<br>
```<br>
ifconfig eth0 | grep 'inet'<br>
## or,<br>
#ip addr show dev eth0 | grep 'inet'<br>
## followed by<br>
curl '<a href="http://ipinfo.io/ip" target="_blank">ipinfo.io/ip</a>'<br>
```<br>
<br>
You may run RTP-Proxy as:<br>
<br>
```<br>
sudo rtpproxy -d DBUG:LOG_LOCAL5 -f -s udp:<a href="http://127.0.0.1:12221" target="_blank">127.0.0.1:12221</a> -u<br>
rtpproxy:rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -A 54.111.222.123 -l<br>
10.0.0.2 -m 16384 -M 32768<br>
```<br>
<br>
[0] <a href="http://bit.ly/1KUS9bx" target="_blank">http://bit.ly/1KUS9bx</a><br>
[1] <a href="https://github.com/sippy/rtpproxy" target="_blank">https://github.com/sippy/rtpproxy</a><br>
[2] <a href="http://bit.ly/1CpQXe9" target="_blank">http://bit.ly/1CpQXe9</a><span><br>
<br>
</span><span><font color="#888888">--<br>
Regards,<br>
Babil (Golam Sarwar)<br>
<br>
PGP Key Fingerprint : D3A1 EED0 5BA0 72D3 A011 75CB 8EA6 7D99 F433 E92D<br>
PGP Key Download URL: <a href="http://bit.ly/gsbabil-pgp-key" target="_blank">http://bit.ly/gsbabil-pgp-key</a></font></span></div><div class="gmail_extra"><br clear="all"><div><div>Regards,<br>Babil (Golam Sarwar)<br><br>PGP Key Fingerprint : D3A1 EED0 5BA0 72D3 A011 75CB 8EA6 7D99 F433 E92D<br>PGP Key Download URL: <a href="http://bit.ly/gsbabil-pgp-key" target="_blank">http://bit.ly/gsbabil-pgp-key</a></div></div>
<br><div class="gmail_quote"><div><div class="h5">On Fri, Feb 13, 2015 at 10:27 AM, Semen Golubcov <span dir="ltr"><<a href="mailto:thegolub4@gmail.com" target="_blank">thegolub4@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 class="h5"><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:ii_i63wh3hr0_14b842b40226eeab" 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:ii_i63wj7qq1_14b842cc331ba96a" 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:ii_i63wk7972_14b842d76c419fe7" 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>
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<br></span></blockquote></div><br></div>
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<br></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature"><div dir="ltr"><div><span style="color:rgb(102,102,102)"><br>Regards,</span></div>
<div><span style="color:rgb(102,102,102)">John Mathew</span></div>
<div><span style="color:rgb(102,102,102)">CEO/Director</span><br><span style="color:rgb(255,102,0)">Divox</span> <span style="color:rgb(102,102,102)">International Inc.</span><br><font color="#666666">Contact: +91-9037100001<br>Email/MSN: <a href="mailto:John.Mathew@divoxmedia.com" target="_blank">John.Mathew@divoxmedia.com</a><br>WEB: <a href="http://www.divoxmedia.com/" target="_blank">www.divoxmedia.com</a><br></font><span style="color:rgb(102,102,102)"></span><br></div>
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