<div dir="ltr"><div>Hi Anothony,<br></div><div><br></div><div>Inbound calling is proving to be more challenging. Here's what I've tried from <a href="https://wiki.freeswitch.org/wiki/Auto_NAT">https://wiki.freeswitch.org/wiki/Auto_NAT</a>:</div>
<div><br></div><div>1. Verified the following params were set in my sip profiles (these were already set by default):</div><div><br></div><div><pre style="font-family:monospace,Courier;padding:1em;border:1px dashed rgb(47,111,171);color:rgb(0,0,0);background-color:rgb(249,249,249);line-height:1.3em;font-size:13px">
<param name="ext-rtp-ip" value="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/></pre></div><div class="gmail_extra"><br></div><div class="gmail_extra">2. Ran sofia status, which shows this:</div><div class="gmail_extra"><br></div>
<div class="gmail_extra"><div class="gmail_extra"><br></div><div class="gmail_extra"> Name<span class="" style="white-space:pre"> </span> Type<span class="" style="white-space:pre"> </span> Data<span class="" style="white-space:pre"> </span>State</div>
<div class="gmail_extra">=================================================================================================</div><div class="gmail_extra"> 10.166.245.111<span class="" style="white-space:pre"> </span> alias<span class="" style="white-space:pre"> </span> internal<span class="" style="white-space:pre"> </span>ALIASED</div>
<div class="gmail_extra"> internal-ipv6<span class="" style="white-space:pre"> </span>profile<span class="" style="white-space:pre"> </span> sip:mod_sofia@[::1]:5060<span class="" style="white-space:pre"> </span>RUNNING (0)</div>
<div class="gmail_extra"> external<span class="" style="white-space:pre"> </span>profile<span class="" style="white-space:pre"> </span> <a href="http://sip:mod_sofia@10.166.245.111:5080">sip:mod_sofia@10.166.245.111:5080</a><span class="" style="white-space:pre"> </span>RUNNING (0)</div>
<div class="gmail_extra"> external::<a href="http://example.com">example.com</a><span class="" style="white-space:pre"> </span>gateway<span class="" style="white-space:pre"> </span> <a href="mailto:sip%3Ajoeuser@example.com">sip:joeuser@example.com</a><span class="" style="white-space:pre"> </span>NOREG</div>
<div class="gmail_extra"> internal<span class="" style="white-space:pre"> </span>profile<span class="" style="white-space:pre"> </span> <a href="http://sip:mod_sofia@10.166.245.111:5060">sip:mod_sofia@10.166.245.111:5060</a><span class="" style="white-space:pre"> </span>RUNNING (0)</div>
<div class="gmail_extra"> internal::<a href="http://bandwidth.com">bandwidth.com</a><span class="" style="white-space:pre"> </span>gateway<span class="" style="white-space:pre"> </span> sip:your user <a href="mailto:name@67.231.8.195">name@67.231.8.195</a><span class="" style="white-space:pre"> </span>NOREG</div>
<div class="gmail_extra">=================================================================================================</div><div class="gmail_extra">3 profiles 1 alias</div><div><br></div></div><div class="gmail_extra">
<br></div><div class="gmail_extra">3. Ran <b style="font-size:13px;color:rgb(0,0,0);font-family:sans-serif;line-height:19.1875px">sofia status profile internal</b><span style="font-size:13px;color:rgb(0,0,0);font-family:sans-serif;line-height:19.1875px"> expecting to see ext-rtp-ip and ext-sip-ip set, but they're missing:</span></div>
<div class="gmail_extra"><span style="font-size:13px;color:rgb(0,0,0);font-family:sans-serif;line-height:19.1875px"><br></span></div><div class="gmail_extra"><div class="gmail_extra">Name <span class="" style="white-space:pre"> </span>internal</div>
<div class="gmail_extra">Domain Name <span class="" style="white-space:pre"> </span>N/A</div><div class="gmail_extra">Auto-NAT <span class="" style="white-space:pre"> </span>false</div><div class="gmail_extra">
DBName <span class="" style="white-space:pre"> </span>sofia_reg_internal</div><div class="gmail_extra">Pres Hosts <span class="" style="white-space:pre"> </span>10.166.245.111,10.166.245.111</div><div class="gmail_extra">
Dialplan <span class="" style="white-space:pre"> </span>XML</div><div class="gmail_extra">Context <span class="" style="white-space:pre"> </span>public</div><div class="gmail_extra">Challenge Realm <span class="" style="white-space:pre"> </span>auto_from</div>
<div class="gmail_extra">RTP-IP <span class="" style="white-space:pre"> </span>10.166.245.111</div><div class="gmail_extra">SIP-IP <span class="" style="white-space:pre"> </span>10.166.245.111</div><div class="gmail_extra">
URL <span class="" style="white-space:pre"> </span><a href="http://sip:mod_sofia@10.166.245.111:5060">sip:mod_sofia@10.166.245.111:5060</a></div><div class="gmail_extra">BIND-URL <span class="" style="white-space:pre"> </span>sip:mod_sofia@10.166.245.111:5060;transport=udp,tcp</div>
<div class="gmail_extra">HOLD-MUSIC <span class="" style="white-space:pre"> </span>local_stream://moh</div><div class="gmail_extra">OUTBOUND-PROXY <span class="" style="white-space:pre"> </span>N/A</div><div class="gmail_extra">
CODECS IN <span class="" style="white-space:pre"> </span>G722,PCMU,PCMA,GSM</div><div class="gmail_extra">CODECS OUT <span class="" style="white-space:pre"> </span>G722,PCMU,PCMA,GSM</div><div class="gmail_extra">
TEL-EVENT <span class="" style="white-space:pre"> </span>101</div><div class="gmail_extra">DTMF-MODE <span class="" style="white-space:pre"> </span>rfc2833</div><div class="gmail_extra">CNG <span class="" style="white-space:pre"> </span>13</div>
<div class="gmail_extra">SESSION-TO <span class="" style="white-space:pre"> </span>0</div><div class="gmail_extra">MAX-DIALOG <span class="" style="white-space:pre"> </span>0</div><div class="gmail_extra">NOMEDIA <span class="" style="white-space:pre"> </span>false</div>
<div class="gmail_extra">LATE-NEG <span class="" style="white-space:pre"> </span>true</div><div class="gmail_extra">PROXY-MEDIA <span class="" style="white-space:pre"> </span>false</div><div class="gmail_extra">
ZRTP-PASSTHRU <span class="" style="white-space:pre"> </span>true</div><div class="gmail_extra">AGGRESSIVENAT <span class="" style="white-space:pre"> </span>false</div><div class="gmail_extra">CALLS-IN <span class="" style="white-space:pre"> </span>86</div>
<div class="gmail_extra">FAILED-CALLS-IN <span class="" style="white-space:pre"> </span>86</div><div class="gmail_extra">CALLS-OUT <span class="" style="white-space:pre"> </span>0</div><div class="gmail_extra">FAILED-CALLS-OUT <span class="" style="white-space:pre"> </span>0</div>
<div class="gmail_extra">REGISTRATIONS <span class="" style="white-space:pre"> </span>1</div><div><br></div><div><br></div><div>So regarding NAT issues, it seems Freeswitch isn't able to get the server's public IP perhaps, despite auto-nat being the correct setting? This is just an EC2 server behind Amazon NAT.</div>
</div><div class="gmail_extra"><span style="font-size:13px;color:rgb(0,0,0);font-family:sans-serif;line-height:19.1875px"><br></span></div><div class="gmail_extra"><span style="font-size:13px;color:rgb(0,0,0);font-family:sans-serif;line-height:19.1875px">Also, I ran:</span></div>
<div class="gmail_extra"><div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px">nat_map status</span></font></div><div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px"><br>
</span></font></div><div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px">Nat Type: UNKNOWN, ExtIP: </span></font></div><div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px">NAT port mapping enabled.</span></font></div>
<div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px"><br></span></font></div><div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px">0 total.</span></font></div>
<div style="color:rgb(0,0,0);font-family:sans-serif;font-size:13px;line-height:19.1875px"><br></div></div><div class="gmail_extra"><br></div><div class="gmail_extra">I did perform an echo test from user 1000 to 9196 from Chrome to Freeswitch, which works as expected, but getting Bandwidth incoming calls to be processed correctly is a challenge. It just loops over and over again with this in the DEBUG console:</div>
<div class="gmail_extra"><br></div><div class="gmail_extra"><div class="gmail_extra">2013-09-13 20:27:31.255055 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/+<a href="mailto:19712383780@192.168.37.68">19712383780@192.168.37.68</a> [80196f26-febf-4363-b48c-ee88022c6b9c]</div>
<div class="gmail_extra">2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+<a href="mailto:19712383780@192.168.37.68">19712383780@192.168.37.68</a> [BREAK]</div><div class="gmail_extra">
2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+<a href="mailto:19712383780@192.168.37.68">19712383780@192.168.37.68</a> [BREAK]</div><div class="gmail_extra">2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/+<a href="mailto:19712383780@192.168.37.68">19712383780@192.168.37.68</a>) Running State Change CS_NEW</div>
<div class="gmail_extra">2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/+<a href="mailto:19712383780@192.168.37.68">19712383780@192.168.37.68</a>) State NEW</div><div class="gmail_extra">
2013-09-13 20:27:31.275057 [DEBUG] sofia.c:8003 IP 67.231.4.195 Rejected by acl "domains". Falling back to Digest auth.</div><div class="gmail_extra">2013-09-13 20:27:31.275057 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+<a href="mailto:19712383780@192.168.37.68">19712383780@192.168.37.68</a> [BREAK]</div>
<div class="gmail_extra">2013-09-13 20:27:31.275057 [DEBUG] sofia.c:1787 detaching session 80196f26-febf-4363-b48c-ee88022c6b9c</div><div><br></div></div><div class="gmail_extra"><br></div><div class="gmail_extra">So in summary, the WebRTC seems good but not the PSTN. I can do Chrome to FS to Chrome, Chrome to FS Echo, I can call from Chrome to FS to Bandwidth (NO AUDIO) but calling inbound loops. I'm assuming all of the routing is done using the Bandwidth DID I've pointed to the server, not the 971xxxxxxx number that I'm calling from on Verizon/Google Voice.</div>
<div class="gmail_extra"><br></div><div class="gmail_extra">Thank you for your help!</div><div class="gmail_extra">James</div><div class="gmail_extra"><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Sep 13, 2013 at 11:51 AM, 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:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">That should not matter. It will be taken care of.<div>
<br></div><div>Verify you can just use bandwidth to call the server and run an echo test or something.</div><div>Then try a sip phone registered. Then on to the webrtc instance.</div>
<div><br></div><div>You probably have some nat problems to bandwidth regardless of WebRTC.</div></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div class="h5">On Fri, Sep 13, 2013 at 1:42 PM, James Mortensen <span dir="ltr"><<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div class="h5"><div dir="ltr">Here is my follow up to this issue. It seems something happened to the server where it stopped sending Binding responses to Chrome. This is a known issue that I've seen before that Google says was an Asterisk issue. However, the same thing happens to Freeswitch as well, indicating the problem is the network/server, not the software.<div>
<br></div><div>I booted up another server from a snapshot, verified two way audio with Asterisk, then reinstalled Freeswitch, uncommented the ws-binding parameter to enable WS on port 5066, then registered user 1000 and 1002 from the <a href="http://tryit.jssip.net" target="_blank">tryit.jssip.net</a> demo using the following configuration:</div>
<div><br></div><div>Name: James</div><div>SIP URI: sip:1000@54.X.X.X <-- public IP of server</div><div>password: 1234</div><div>WS URI: ws://54.X.X.X:5066 <--- same public IP</div><div><br></div><div>and I substituted 1002 in place of 1000 for another user on another network. I verified audio flows both ways with audio flowing through Freeswitch.</div>
<div><br></div><div>I made no other changes to the configuration. This was a lot easier to get started with than Asterisk WebRTC. I just made it out to be a lot harder than it was by assuming I needed to manually add in my server's IP address in the configuration files. This happens automatically, even on a NAT'd server. Amazing! :)</div>
<div><br></div><div><br></div><div>Now, for the PSTN part, I configured my BANDWIDTH.com provider as an internal context and as an outbound and inbound dialplan. I can connect a call between my cellphone and Chrome, but there's no audio flowing to/from Bandwidth. I suspect the problem has something to do with bridging AVP and SAVPF, since the carriers don't support SRTP.</div>
<div><br></div><div>I've grepped the configuration files, and I don't see how I would configure the system to bridge AVP and SAVPF and do transcoding. Any ideas?</div><div><br></div><div>Thank you!!</div><span><font color="#888888"><div>
<br>
</div><div>James</div></font></span><div><div><div class="gmail_extra"><div><div dir="ltr"><div><br></div><div><br></div></div></div><br><div class="gmail_quote">On Wed, Sep 11, 2013 at 12:01 PM, James Mortensen <span dir="ltr"><<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr"><div class="gmail_extra"><div><div dir="ltr">
<div>Installing <span style="line-height:17px;font-size:13px;background-color:rgb(240,240,240);font-family:Arial,FreeSans,Helvetica,sans-serif">ibncursesw5 and </span><span style="line-height:17px;font-size:13px;background-color:rgb(240,240,240);font-family:Arial,FreeSans,Helvetica,sans-serif">libncursesw5-dev</span> did resolve the issue. I can now connect to the WebSocket server. Seems the cluechoo module is just something that was added in to separate the help vampires from the people with legitimate issues. :D</div>
<div><br></div><div>I got 405 Method Not Allowed errors, and I resolved them by making sure the IP address in the SIP URI matches the IP address in the WS field. </div><div><br></div><div>At this time, I've successfully registered and am getting back 200 OK's and have now moved onto tweaking the settings so I can get audio going both ways. The candidates appear to be only showing the public IP, so I'll have to figure that out. I'll provide more updates, or questions, as I move forward.</div>
<span><font color="#888888">
<div><br></div><div>James</div></font></span></div></div>
<br><br><div class="gmail_quote"><div>On Wed, Sep 11, 2013 at 11:32 AM, James Mortensen <span dir="ltr"><<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>></span> wrote:<br>
</div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div dir="ltr"><div>Okay, I added in the dependencies and still get those errors. Do I need to file a bug in JIRA for this or am I just missing something? <div><br></div><div>I'm happy to try a different OS if there's one that's been tried and tested.</div>
<div><br></div><div>Of course, more googling reveals that mod_cluechoo is just a joke? <a href="http://wiki.freeswitch.org/wiki/Mod_cluechoo" target="_blank">http://wiki.freeswitch.org/wiki/Mod_cluechoo</a></div><div><span style="font-size:medium;font-family:Times"><br>
</span></div><div><span style="font-size:medium;font-family:Times">> SL (Steam Locomotive) runs across your terminal when you type "sl" as you meant to type "ls". It's just a joke command, and not usefull at all. Put the binary to /usr/local/bin.</span></div>
<div><br></div><div>I hope I'm not chasing a problem that has absolutely no impact on my ability to get WebRTC working here. :D</div><div><br></div><div><br></div><div>Thanks for any additional help you can provide,</div>
<div>James</div></div><div><div><div><br></div><div class="gmail_extra"><div><div dir="ltr"><div><br></div></div></div><br><div class="gmail_quote"><div>On Wed, Sep 11, 2013 at 11:19 AM, James Mortensen <span dir="ltr"><<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>></span> wrote:<br>
</div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div dir="ltr">I ran the netstat command, and it doesn't appear to be listening. It doesn't appear to be listening on anything. I do have this error in the console when starting:<div>
<br></div><div><div>2013-09-11 18:16:14.445921 [ERR] switch_nat.c:201 Error checking for PMP [general error]</div>
<div><br></div><div>AND</div><div><br></div><div><div>2013-09-11 18:09:30.233568 [CRIT] switch_loadable_module.c:1383 Error Loading module /opt/freeswitch-1.4b/mod/mod_cluechoo.so</div><div>**/opt/freeswitch-1.4b/mod/mod_cluechoo.so: undefined symbol: waddch**</div>
</div><div><br></div><div>I believe I overlooked these earlier. But it's possible I'm missing a dependency. I'm going to install the libncurses packages as described here <a href="http://jira.freeswitch.org/browse/FS-3689" target="_blank">http://jira.freeswitch.org/browse/FS-3689</a> and then rebuild to see if that helps. I'm running Ubuntu 12.10.</div>
<span><font color="#888888">
</font></span></div><span><font color="#888888"><div><br></div><div>James</div><div><br></div></font></span></div></div><div class="gmail_extra"><div><div><div dir="ltr"><div>
</div></div></div>
<br><br></div><div><div><div><div class="gmail_quote">On Wed, Sep 11, 2013 at 11:07 AM, 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:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div dir="ltr">Did you open all the necessary firewall ports?<div>Playing around on amazon as your first try complicates thing a bit for you.</div><div><br></div><div>You should be able to verify its listening on the port with netstat -an | grep 5066</div>
<div><br></div></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Sep 11, 2013 at 12:55 PM, James Mortensen <span dir="ltr"><<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">Here's another update to my adventures in Freeswitch WebRTC. I assume from the getting started documentation that there are users created by default with the password 1234, so I'm trying to create the ws 5066 connection from the TryIt JsSIP demo: <a href="http://tryit.jssip.net" target="_blank">http://tryit.jssip.net</a><div>
<br></div><div>Name: James</div><div>SIP URI: 1000@Y.Y.Y.Y <--- Local IP of EC2 server<br><div class="gmail_extra">SIP password: 1234</div><div class="gmail_extra">WS URI: ws://X.X.X.X:5066 <--- Public IP of EC2 server<br clear="all">
<div><div dir="ltr"><br></div></div><div>Hope this helps!</div><span><font color="#888888"><div>James</div></font></span><div><div>
<br><br><div class="gmail_quote"><br></div></div></div></div></div></div></blockquote></div></div></div></div></blockquote></div><br></div></div></div></div>
</blockquote></div><br></div></div></div></div>
</blockquote></div><br></div></div>
</blockquote></div><br></div></div></div></div>
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<br></div></div></blockquote></div><div><div class="h5"><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
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<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
<a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
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