Hi Anothony, Inbound calling is proving to be more challenging. Here's what I've tried from https://wiki.freeswitch.org/wiki/Auto_NAT: 1. Verified the following params were set in my sip profiles (these were already set by default): <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> 2. Ran sofia status, which shows this: Name Type Data State ================================================================================================= 10.166.245.111 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia@10.166.245.111:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG internal profile sip:mod_sofia@10.166.245.111:5060 RUNNING (0) internal::bandwidth.com gateway sip:your user name@67.231.8.195 NOREG ================================================================================================= 3 profiles 1 alias 3. Ran *sofia status profile internal* expecting to see ext-rtp-ip and ext-sip-ip set, but they're missing: Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 10.166.245.111,10.166.245.111 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.166.245.111 SIP-IP 10.166.245.111 URL sip:mod_sofia@10.166.245.111:5060 BIND-URL sip:mod_sofia@10.166.245.111:5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 86 FAILED-CALLS-IN 86 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1 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. Also, I ran: nat_map status Nat Type: UNKNOWN, ExtIP: NAT port mapping enabled. 0 total. 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: 2013-09-13 20:27:31.255055 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/+19712383780@192.168.37.68[80196f26-febf-4363-b48c-ee88022c6b9c] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/+19712383780@192.168.37.68) Running State Change CS_NEW 2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/+19712383780@192.168.37.68) State NEW 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. 2013-09-13 20:27:31.275057 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+19712383780@192.168.37.68 [BREAK] 2013-09-13 20:27:31.275057 [DEBUG] sofia.c:1787 detaching session 80196f26-febf-4363-b48c-ee88022c6b9c 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. Thank you for your help! James On Fri, Sep 13, 2013 at 11:51 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
That should not matter. It will be taken care of.
Verify you can just use bandwidth to call the server and run an echo test or something. Then try a sip phone registered. Then on to the webrtc instance.
You probably have some nat problems to bandwidth regardless of WebRTC.
On Fri, Sep 13, 2013 at 1:42 PM, James Mortensen < james.mortensen@synclio.com> wrote:
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.
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 tryit.jssip.net demo using the following configuration:
Name: James SIP URI: sip:1000@54.X.X.X <-- public IP of server password: 1234 WS URI: ws://54.X.X.X:5066 <--- same public IP
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.
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! :)
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.
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?
Thank you!!
James
On Wed, Sep 11, 2013 at 12:01 PM, James Mortensen < james.mortensen@synclio.com> wrote:
Installing ibncursesw5 and libncursesw5-dev 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
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.
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.
James
On Wed, Sep 11, 2013 at 11:32 AM, James Mortensen < james.mortensen@synclio.com> wrote:
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?
I'm happy to try a different OS if there's one that's been tried and tested.
Of course, more googling reveals that mod_cluechoo is just a joke? http://wiki.freeswitch.org/wiki/Mod_cluechoo
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.
I hope I'm not chasing a problem that has absolutely no impact on my ability to get WebRTC working here. :D
Thanks for any additional help you can provide, James
On Wed, Sep 11, 2013 at 11:19 AM, James Mortensen < james.mortensen@synclio.com> wrote:
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:
2013-09-11 18:16:14.445921 [ERR] switch_nat.c:201 Error checking for PMP [general error]
AND
2013-09-11 18:09:30.233568 [CRIT] switch_loadable_module.c:1383 Error Loading module /opt/freeswitch-1.4b/mod/mod_cluechoo.so **/opt/freeswitch-1.4b/mod/mod_cluechoo.so: undefined symbol: waddch**
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 http://jira.freeswitch.org/browse/FS-3689 and then rebuild to see if that helps. I'm running Ubuntu 12.10.
James
On Wed, Sep 11, 2013 at 11:07 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Did you open all the necessary firewall ports? Playing around on amazon as your first try complicates thing a bit for you.
You should be able to verify its listening on the port with netstat -an | grep 5066
On Wed, Sep 11, 2013 at 12:55 PM, James Mortensen < james.mortensen@synclio.com> wrote:
> 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: http://tryit.jssip.net > > Name: James > SIP URI: 1000@Y.Y.Y.Y <--- Local IP of EC2 server > SIP password: 1234 > WS URI: ws://X.X.X.X:5066 <--- Public IP of EC2 server > > Hope this helps! > James > > > >
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