-
Notifications
You must be signed in to change notification settings - Fork 2.4k
New issue
Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.
By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.
Already on GitHub? Sign in to your account
[ SIP Gateway ] register failed. #11
Comments
This looks like a sofia-sip issue: I guess something on your machine is preventing it to bind to a port for the SIP client. I'll have to check whether there's a way to force it to stop after a number of attempts, rather than trying indefinitely. Do you have anything installed that may interfere? The only related issue I could find around was on SElinux that could cause something like this. If so, you may try disabling it temporarily to see if that fixes it, and in case then try to prepare a policy for it rather than disabling it entirely. |
I have been able to register devices to Asterisk via the Janus-gateway and I am using Ubuntu 12.04 and Asterisk 11.8.0, with the vp8/opus patch Hope this helps you get going, if not post question(s) and I will be glad Cheers Config files: http.conf rtp.conf sip.conf [authentication] [8000] ;chrome browser windows 7 [8001] ;chromium linux browser On Wed, Mar 19, 2014 at 3:24 AM, ducdung0909 notifications@github.comwrote:
|
Hi, @tchandler48 : Here is my asterisk configuration. Should I adjust anything ? |
I don't think Asterisk has anything to do with this. The problem lies within the SIP stack the plugin uses even before getting to Asterisk at all. What is the output of the console when the gateway is started? I'm especially thinking about the output when the SIP plugin is first started, as it may display information on the cause of the issue. |
Hi, What release of Asterisk are you running? (11.x.x, etc) I would add in the I would change type=peer to type=friend if running Asterisk 11.x.x, and you want video, you must apply the So If, I understand your issue, the sip gateway web page will not register Cheers On Wed, Mar 19, 2014 at 11:11 PM, ducdung0909 notifications@github.comwrote:
|
In my testing of the sip plugin, registration was not a problem. The first Cheers On Thu, Mar 20, 2014 at 5:03 AM, Lorenzo Miniero
|
@tchandler48 : I can't make register for an user, I haven't made a call yet. Starting Meetecho Janus (WebRTC Gateway)Reading configuration from ./conf/janus.cfg |
From the log I can only see that, for what concerns interfaces, the gateway is using 192.168.16.60, while 192.168.122.1 is used by the SIP plugin. Was this done on purpose, e.g., via configuration? Do both the interfaces work fine? Anyway, that should not be the issue here, as that autodetection is only used to populate the IP addresses in SIP and SDP accordingly. Can you try building and running the Sofia-SIP test applications that you can find here, and check whether they work fine? Just to try and understand where the specific issue might be:
This will build two small applications, a caller and callee, that you can use to test both sides of a SIP session using the stack. The only difference when creating the NUA between the tests and my plugin is that the test application binds to a specific port (5060 for the callee, 5062 for the caller) for all interfaces (0.0.0.0), while the plugin uses a wildcard for the port instead (0.0.0.0:*). Keep me posted! |
Dear Iminero, |
Hi, [Mar 24 10:30:43] WARNING[3994][C-00000000]: chan_sip.c:10124 process_sdp: Received AVP profile in audio offer but AVPF is enabled: audio 17194 RTP/AVP 111 103 104 0 8 106 105 13 126 I think I should install newest version of Asterisk ( my current version is 11.5 without any patch ). |
I guess that was the reason then: maybe sofia-sip doesn't like virtual interfaces? I'll check if there's an easy way to skip them, and I'll add a way to instruct the plugin to only bind on specific interfaces rather than all of them. For what concerns the Asterisk error, no need to update anything. I guess you just have a avpf=yes in your sip.conf, that when enabled rejects all SDPs that are just AVP instead. If it's not there for a specific reason, you may want to comment that or turn it into an avpf=no; otherwise, try changing the code in line 673 of the SIP plugin, so that the RTP/SAVPFis turned into an RTP/AVPF rather than RTP/AVP, which should make your Asterisk happy. |
Hi, |
That's a problem I noticed as well. Specifically, even when the caller only negotiates audio, if video is enabled on Asterisk the SDP for the callee has a video line as well. In case the callee supports video this means it is negotiated as well, despite the fact that the caller never asked for it and is not going to do it. This is why you're getting the black video on the callee side: you have a video element, but no video frames. Can you at least confirm that calling generic default extensions on Asterisk (e.g., echo test, confbridge, any DTMF-based menu, etc.) works fine? Apart from that, there still are issues @tchandler48 documented in a different issue, and that I hope I'll be able to look into ASAP. |
Try disabling video on the Asterisk side (in sip.conf, video=no) to also check whether or not you're able to get audio calls to work that way. |
Hi, Now, the call is established and connected but still not heard anything at bothside. == Using SIP RTP CoS mark 5 and output of gateway: Request completed, freeing data |
Sorry, but I see I get one more issue. When I hangup the call, at both points ( caller and callee ), the button "hangup" switch to blur effect, they don't switch back to call button for making another calls. The call doesn't hangup completely. |
I am traveling, but will look at this tonight. I see a couple of things Tom C. On Mon, Mar 24, 2014 at 10:54 PM, ducdung0909 notifications@github.comwrote:
|
@ducdung0909 I just updated the SIP plugin, could you let me known if the new version still gives you the same issues? Please notice that the syntax for registering and calling has changed a bit to make the plugin more generic and usable with other SIP servers as well (e.g., Kamailio). |
Dear lminiero, ..... For more information, I used same configuration content of janus. Asterisk or Kamailio is normal, successfully registering and calling by using softphone ( zoiper, xlite, jitsi ...). |
Hi, As I said to Tom in the other issue, the web page and JavaScript code for Lorenzo
|
Dear @lminiero , There's a message for JANUS SIP plugin |
I know you updated the gateway :-) What I meant is that the demo pages that interface to it have changed as well. The error you're getting (missing proxy) means that the web page (siptest.html+siptest.js) are not sending any proxy field in the JSON request. The updated SIP plugin asks for a proxy value, while the old one used a proxy_ip and proxy_port separated pieces of info, which makes me think you're still using the old siptest.html/.js demos to contact the gateway, and that would explain the issue. Make sure you're using the updated web demos and keep me posted. |
Dear @lminiero , |
You're right, good catch, I think that by default, when answering, siptest.js is requesting accesso to both audio and video no matter what is being negotiated, while we should first look at the offer we got and chech whether both audio and video were requested. I'll add this check in the next update and keep you posted. |
Hi, Name/username Host but
|
I just pushed a fix for the getUserMedia thing: just replace the siptest.js you have with the new one, and let me know if that works for you now. PS: make sure the updated version is loaded, clearing the browser cache might help there. About sip show peers, yes, that's normal. In fact, with Janus involved the actual SIP client is Janus itself, and not the web browser: the Janus SIP plugin acts as a SIP client on behalf of the browser, and so that's why it's the Janus IP that appears when looking at registered peers. |
Hi @lminiero , |
Ok closing the issue then. |
Hi,
I got demo Video Call, Video MCU successfully but I still stucked at SIP Gateway feature. I can't register a sip user to Asterisk Sip Server. Log of Janus gives thounsand of lines and It doesn't stop unless I quit janus.
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 9303, trying 12032
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 12032, trying 14761
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 14761, trying 17490
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 17490, trying 20219
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 20219, trying 22948
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 22948, trying 25677
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 25677, trying 28406
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 28406, trying 31135
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 31135, trying 33864
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 33864, trying 36593
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 36593, trying 39322
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 39322, trying 42051
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 42051, trying 44780
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 44780, trying 47509
tport_bind_server(0x7f7b842ad0d0): cannot bind all transports to port 47509, trying 50238
............................................
What should I do to resolve this issue ? Thanks !
The text was updated successfully, but these errors were encountered: