[ SIP Gateway ] register failed. about janus-gateway

[lminiero]

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.

[tchandler48]

I have been able to register devices to Asterisk via the Janus-gateway and
complete calls. (there are some issues that are being investigated, but it
does work somewhat)...

I am using Ubuntu 12.04 and Asterisk 11.8.0, with the vp8/opus patch
applied. I have listed below the configuration files that I had to change

Hope this helps you get going, if not post question(s) and I will be glad
to help.

Cheers
Tom C.

Config files:

http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088

rtp.conf
[general]
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=stun.l.google.com:19302

sip.conf
[general]
allowoverlap=no
realm=192.168.1.171 (your ip address)
udpbindaddr=0.0.0.0:5060
transport=udp,ws
videosupport=always
nat=force_rport,comedia
encryption=yes
avpf=yes

[authentication]
[2250] ;ata device
type=friend
secret=1234
host=dynamic
context=local
qualify=yes
transport=udp
directmedia=no
videosupport=no
disallow=all
allow=ulaw

[8000] ;chrome browser windows 7
secret=1234
context=local
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
transport=udp,ws
callcounter=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw,vp8

[8001] ;chromium linux browser
secret=1234
context=local
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
transport=ws
callcounter=yes
icesupport=yes
directmedia=no
transport=udp,ws
disallow=all
allow=ulaw,vp8

On Wed, Mar 19, 2014 at 3:24 AM, ducdung0909 [email protected]:

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 !

Reply to this email directly or view it on GitHubhttps://github.com//issues/11
.

[ducdung0909]

Hi,
@lminiero I have set disabled on SELinux before.

@tchandler48 : Here is my asterisk configuration.
htttp.conf and rtp.conf is same as yours.
sip.conf
[1000]
type=peer
username=1000
host=dynamic
secret=1000
context=local1xxx
hasiax = no
hassip = yes
encryption = yes
avpf = yes
icesupport = yes
videosupport=yes
directmedia=no

Should I adjust anything ?

[lminiero]

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.

[tchandler48]

Hi,

What release of Asterisk are you running? (11.x.x, etc)

I would add in the
[1000]
disallow=all
allow=ulaw,vp8

I would change type=peer to type=friend

if running Asterisk 11.x.x, and you want video, you must apply the
patch I attached to my earlier message.

So If, I understand your issue, the sip gateway web page will not register
to the Asterisk?

Cheers
Tom C

On Wed, Mar 19, 2014 at 11:11 PM, ducdung0909 [email protected]:

Hi,
@lminiero https://github.com/lminiero I have set disabled on SELinux
before.

@tchandler48 https://github.com/tchandler48 : Here is my asterisk
configuration.
htttp.conf and rtp.conf is same as yours.
sip.conf
[1000]
type=peer
username=1000
host=dynamic
secret=1000
context=local1xxx
hasiax = no
hassip = yes
encryption = yes
avpf = yes
icesupport = yes
videosupport=yes
directmedia=no

Should I have adjust anything ?

Reply to this email directly or view it on GitHubhttps://github.com//issues/11#issuecomment-38133808
.

[tchandler48]

In my testing of the sip plugin, registration was not a problem. The first
time I tried it, it registered to Asterisk. I later had issues on
connections between different end points, but registering with Asterisk was
painless......

Cheers
Tom C

On Thu, Mar 20, 2014 at 5:03 AM, Lorenzo Miniero
[email protected]:

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.

Reply to this email directly or view it on GitHubhttps://github.com//issues/11#issuecomment-38150529
.

[ducdung0909]

@tchandler48 : I can't make register for an user, I haven't made a call yet.
@lminiero : Here is the output of console when gate start.

---

Starting Meetecho Janus (WebRTC Gateway)

Reading configuration from ./conf/janus.cfg
[janus.cfg]
[general]
configs_folder: ./conf
plugins_folder: ./plugins
[webserver]
http: yes
port: 8088
https: no
secure_port: 8889
base_path: /janus
[certificates]
cert_pem: certs/mycert.pem
cert_key: certs/mycert.key
Checking command line arguments...
[janus.cfg]
[general]
configs_folder: ./conf
plugins_folder: ./plugins
[webserver]
http: yes
port: 8088
https: no
secure_port: 8889
base_path: /janus
[certificates]
cert_pem: certs/mycert.pem
cert_key: certs/mycert.key
Available interfaces:
lo: 127.0.0.1
eth0: 192.168.16.60
br0: 192.168.16.60
virbr0: 192.168.122.1
Using 192.168.16.60 as local IP...
Using certificates:
certs/mycert.pem
certs/mycert.key
Fingerprint of our certificate is C5:5F:DA:7D:84:47:B1:BF:6B:55:16:62:48:31:3E:D3:F1:7B:25:89:92:4A:4B:4D:4D:D9:D5:AF:EA:D8:15:44
Plugins folder: ./plugins
Loading plugin 'janus_streaming.so'...
JANUS Streaming plugin created!
Configuration file: ./conf/janus.plugin.streaming.cfg
[janus.plugin.streaming.cfg]
[gstreamer-sample]
type: rtp
id: 1
description: Opus/VP8 live stream coming from gstreamer
audio: yes
video: yes
audioport: 5002
audiopt: 111
audiortpmap: opus/48000/2
videoport: 5004
videopt: 100
videortpmap: VP8/9000
[file-live-sample]
type: live
id: 2
description: a-law file source
filename: ./plugins/streams/radio.alaw
audio: yes
video: no
[file-ondemand-sample]
type: ondemand
id: 3
description: mu-law file source
filename: ./plugins/streams/music.mulaw
audio: yes
video: no
Adding stream 'gstreamer-sample'
Audio enabled, Video enabled
Adding stream 'file-live-sample'
Adding stream 'file-ondemand-sample'
::: [3][file-ondemand-sample] mu-law file source (on demand, file source)
::: [2][file-live-sample] a-law file source (live, file source)
::: [1][gstreamer-sample] Opus/VP8 live stream coming from gstreamer (live, RTP source)
[janus_streaming.c:janus_streaming_filesource_thread:960:] [janus_streaming.c:janus_streaming_relay_thread:1065:] Filesource RTP thread starting...
Opening file source ./plugins/streams/radio.alaw...
Starting relay thread
JANUS Streaming plugin initialized!
Streaming audio file: ./plugins/streams/radio.alaw
Version: 1 (0.0.1)
[janus.plugin.streaming] JANUS Streaming plugin
This is a streaming plugin for Janus, allowing WebRTC peers to watch/listen to pre-recorded files or media generated by gstreamer.
[gstreamer-sample] Audio listener bound to port 5002
Loading plugin 'janus_videoroom.so'...
[gstreamer-sample] Video listener bound to port 5004
JANUS VideoRoom plugin created!
Configuration file: ./conf/janus.plugin.videoroom.cfg
[janus.plugin.videoroom.cfg]
[1234]
description: Demo Room
publishers: 6
bitrate: 128000
fir_freq: 10
Adding video room '1234'
Created videoroom: 1234 (Demo Room)
::: [1234][Demo Room] 128000, max 6 publishers, FIR frequency of 10 seconds
JANUS VideoRoom plugin initialized!
Version: 1 (0.0.1)
[janus.plugin.videoroom] JANUS VideoRoom plugin
This is a plugin implementing a videoconferencing MCU for Janus, something like Licode.
Loading plugin 'janus_videocall.so'...
[janus_videoroom.c:janus_videoroom_handler:538:] Joining thread
JANUS VideoCall plugin created!
Configuration file: ./conf/janus.plugin.videocall.cfg
[janus.plugin.videocall.cfg]
JANUS VideoCall plugin initialized!
Version: 1 (0.0.1)
[janus.plugin.videocall] JANUS VideoCall plugin
[janus_videocall.c:janus_videocall_handler:349:] This is a simple video call plugin for Janus, allowing two WebRTC peers to call each other through the gateway.
Joining thread
Loading plugin 'janus_sip.so'...
JANUS SIP plugin created!
Configuration file: ./conf/janus.plugin.sip.cfg
[janus.plugin.sip.cfg]
Available interfaces:
lo: 127.0.0.1
eth0: 192.168.16.60
br0: 192.168.16.60
[janus_streaming.c:janus_streaming_handler:644:] virbr0: 192.168.122.1
Joining thread
Using 192.168.122.1 as local IP...
JANUS SIP plugin initialized!
Version: 1 (0.0.1)
[janus.plugin.sip] JANUS SIP plugin
[janus_sip.c:janus_sip_handler:491:] This is a simple SIP plugin for Janus, allowing WebRTC peers to register at a SIP server and call SIP user agents through the gateway.
Joining thread
Loading plugin 'janus_voicemail.so'...
JANUS VoiceMail plugin created!
Configuration file: ./conf/janus.plugin.voicemail.cfg
[janus.plugin.voicemail.cfg]
[general]
path: ./html/voicemail/
base: /voicemail/
Recordings path: ./html/voicemail/
Recordings base: /voicemail/
JANUS VoiceMail plugin initialized!
Version: 1 (0.0.1)
[janus.plugin.voicemail] JANUS VoiceMail plugin
This is a plugin implementing a very simple VoiceMail service for Janus, recording Opus streams.
[janus_voicemail.c:janus_voicemail_handler:416:] Loading plugin 'janus_echotest.so'...
Joining thread
JANUS EchoTest plugin created!
Configuration file: ./conf/janus.plugin.echotest.cfg
[janus.plugin.echotest.cfg]
JANUS EchoTest plugin initialized!
Version: 1 (0.0.1)
[janus.plugin.echotest] JANUS EchoTest plugin
This is a trivial EchoTest plugin for Janus, just used to showcase the plugin interface.
Loading plugin 'janus_audiobridge.so'...
[janus_echotest.c:janus_echotest_handler:323:] Joining thread
JANUS AudioBridge plugin created!
Configuration file: ./conf/janus.plugin.audiobridge.cfg
[janus.plugin.audiobridge.cfg]
[1234]
description: Demo Room
sampling_rate: 16000
record: yes
Adding audio room '1234'
Created audiobridge: 1234 (Demo Room)
::: [1234][Demo Room] 16000 (will be recorded)
JANUS AudioBridge plugin initialized!
Version: 1 (0.0.1)
[janus.plugin.audiobridge] JANUS AudioBridge plugin
This is a plugin implementing an audio conference bridge for Janus, mixing Opus streams.
Audio bridge thread starting...
[janus_audiobridge.c:janus_audiobridge_handler:498:] Thread is for mixing room 1234 (Demo Room)...
Joining thread
Recording requested, opened file /tmp/janus-audioroom-1234.wav for writing
HTTP webserver started (port 8088, /janus path listener)...
HTTPS webserver disabled

[lminiero]

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:

https://gitorious.org/sofia-sip/pages/SofiaTutorial

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!

[ducdung0909]

Dear Iminero,
I'm going try to use your recommendation applications for checking. I will reply soon. Thank you.

[ducdung0909]

Hi,
I turned off all virt interface ( br0, virbr0 ) and got successful registeration. But now, I can't make a call and get warning from asterisk :

[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
[Mar 24 10:30:43] WARNING[3994][C-00000000]: chan_sip.c:10464 process_sdp: Insufficient information in SDP (c=)...

I think I should install newest version of Asterisk ( my current version is 11.5 without any patch ).

[lminiero]

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.

[ducdung0909]

Hi,
As your suggestion, I can make call now without any warning of Asterisk. I try to make some noise but I dont hear anything at both side. Moreover, the caller is only asked about using microphone but the callee is asked to use both camera and microphone. I see caller have 2 black screens at video area while callee has local video and black screen of remote video area.

[lminiero]

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.

[lminiero]

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.

[ducdung0909]

Hi,
I remove these configuration in sip.conf at every specific user.
avpf=yes
encryption=yes
videosupport=yes

Now, the call is established and connected but still not heard anything at bothside.
Here is log of asterisk.

== Using SIP RTP CoS mark 5
-- Executing [1001@local1xxx:1] Dial("SIP/1000-00000000", "SIP/1001") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/1001
-- SIP/1001-00000001 is ringing
-- SIP/1001-00000001 answered SIP/1000-00000000
-- Channel SIP/1000-00000000 joined 'simple_bridge' basic-bridge <6b7791d0-0d81-4ac2-a8be-cf24cf4fd1c3>
-- Channel SIP/1001-00000001 joined 'simple_bridge' basic-bridge <6b7791d0-0d81-4ac2-a8be-cf24cf4fd1c3>

and output of gateway:

Request completed, freeing data
Got a HTTP GET request on /janus/1698893131...
... Just parsing headers for now...
Host: 192.168.16.64:8088
Connection: keep-alive
Accept: application/json, text/javascript, /; q=0.01
Origin: http://192.168.16.64
User-Agent: Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/33.0.1750.154 Safari/537.36
Referer: http://192.168.16.64/janus/siptest.html
Accept-Encoding: gzip,deflate,sdch
Accept-Language: en-US,en;q=0.8
Got a HTTP GET request on /janus/1698893131...
... parsing request...
Session: 1698893131
Session 1698893131 found... returning message
... handling long poll...
[3562290375] Got an RTCP packet (audio stream)!
[1062166045] Got an RTCP packet (audio stream)!
[3562290375] Got an RTCP packet (audio stream)!
Long poll time out for session 3086404912...
We have a message to serve...
{"janus" : "keepalive"}
Request completed, freeing data
Got a HTTP GET request on /janus/3086404912...
... Just parsing headers for now...
Host: 192.168.16.64:8088
Connection: keep-alive
Accept: application/json, text/javascript, /; q=0.01
Origin: http://192.168.16.64
User-Agent: Mozilla/5.0 (Windows NT 6.1) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/34.0.1847.76 Safari/537.36
Referer: http://192.168.16.64/janus/siptest.html
Accept-Encoding: gzip,deflate,sdch
Accept-Language: vi,fr-FR;q=0.8,fr;q=0.6,en-US;q=0.4,en;q=0.2,ko;q=0.2,af;q=0.2
Got a HTTP GET request on /janus/3086404912...
... parsing request...
Session: 3086404912
Session 3086404912 found... returning message
... handling long poll...
[1062166045] Got an RTCP packet (audio stream)!
[3562290375] Got an RTCP packet (audio stream)!
[1062166045] Got an RTCP packet (audio stream)!

[ducdung0909]

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.

[tchandler48]

I am traveling, but will look at this tonight. I see a couple of things
and also I need to check the chrome settings.

Tom C.

On Mon, Mar 24, 2014 at 10:54 PM, ducdung0909 [email protected]:

Hi,
I remove these configuration in sip.conf at every specific user.
avpf=yes
encryption=yes
videosupport=yes

Now, the call is established and connected but still not heard anything at
bothside.
Here is log of asterisk.

== Using SIP RTP CoS mark 5
-- Executing [1001@local1xxx:1] Dial("SIP/1000-00000000", "SIP/1001") in
new stack
== Using SIP RTP CoS mark 5
-- Called SIP/1001
-- SIP/1001-00000001 is ringing
-- SIP/1001-00000001 answered SIP/1000-00000000
-- Channel SIP/1000-00000000 joined 'simple_bridge' basic-bridge
-- Channel SIP/1001-00000001 joined 'simple_bridge' basic-bridge

and output of gateway:

Request completed, freeing data
Got a HTTP GET request on /janus/1698893131...
... Just parsing headers for now...
Host: 192.168.16.64:8088
Connection: keep-alive
Accept: application/json, text/javascript, /; q=0.01
Origin: http://192.168.16.64
User-Agent: Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36 (KHTML,
like Gecko) Chrome/33.0.1750.154 Safari/537.36
Referer: http://192.168.16.64/janus/siptest.html
Accept-Encoding: gzip,deflate,sdch
Accept-Language: en-US,en;q=0.8
Got a HTTP GET request on /janus/1698893131...
... parsing request...
Session: 1698893131
Session 1698893131 found... returning message
... handling long poll...
[3562290375] Got an RTCP packet (audio stream)!
[1062166045] Got an RTCP packet (audio stream)!
[3562290375] Got an RTCP packet (audio stream)!
Long poll time out for session 3086404912...
We have a message to serve...
{"janus" : "keepalive"}
Request completed, freeing data
Got a HTTP GET request on /janus/3086404912...
... Just parsing headers for now...
Host: 192.168.16.64:8088
Connection: keep-alive
Accept: application/json, text/javascript, /; q=0.01
Origin: http://192.168.16.64
User-Agent: Mozilla/5.0 (Windows NT 6.1) AppleWebKit/537.36 (KHTML, like
Gecko) Chrome/34.0.1847.76 Safari/537.36
Referer: http://192.168.16.64/janus/siptest.html
Accept-Encoding: gzip,deflate,sdch
Accept-Language:
vi,fr-FR;q=0.8,fr;q=0.6,en-US;q=0.4,en;q=0.2,ko;q=0.2,af;q=0.2
Got a HTTP GET request on /janus/3086404912...
... parsing request...
Session: 3086404912
Session 3086404912 found... returning message
... handling long poll...
[1062166045] Got an RTCP packet (audio stream)!
[3562290375] Got an RTCP packet (audio stream)!
[1062166045] Got an RTCP packet (audio stream)!

Reply to this email directly or view it on GitHubhttps://github.com//issues/11#issuecomment-38528612
.

[lminiero]

@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).

[ducdung0909]

Dear lminiero,
I have tried new source of janus gateway but not successful registering. The "register" button was blured all time and I got the warning as below
#################

.....
Creating new session: 1531118051
Creating new handle in session 1531118051: 825975178
There's a message for JANUS SIP plugin
[ERR] [janus_sip.c:janus_sip_handler:678:] Missing or invalid element (proxy)
#################
Moreover, I switch back to last previous source and got successful registering. But I also got warning below:
#################
.....
There's a message for JANUS SIP plugin
outbound(0x7f17d4001690): FAILED to validate sip:[email protected]:59385;transport=udp
outbound(0x7f17d4001690): FAILED with 404 Not Found
Detaching handle from JANUS SIP plugin
No WebRTC media anymore
Destroying session 2327869492
##################

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 ...).

[lminiero]

Hi,

As I said to Tom in the other issue, the web page and JavaScript code for
the SIP demo have changed, since the syntax used by the plugin were
modified. Are you using the updated ones for interacting with the updated
plugin and gateway?

Lorenzo
Il 16/mag/2014 04:24 "ducdung0909" [email protected] ha scritto:

Dear lminiero,
I have tried new source of janus gateway but not successful registering.
The "register" button was blured all time and I got the warning as below
#################

.....
Creating new session: 1531118051
Creating new handle in session 1531118051: 825975178
There's a message for JANUS SIP plugin
[ERR] [janus_sip.c:janus_sip_handler:678:] Missing or invalid element
(proxy)
#################
Moreover, I switch back to last previous source and got successful
registering. But I also got warning below:
#################
.....
There's a message for JANUS SIP plugin
outbound(0x7f17d4001690): FAILED to validate
sip:[email protected]:59385;transport=udp
outbound(0x7f17d4001690): FAILED with 404 Not Found
Detaching handle from JANUS SIP plugin
No WebRTC media anymore
Destroying session 2327869492
##################

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 ...).

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Reply to this email directly or view it on GitHubhttps://github.com//issues/11#issuecomment-43289092
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[ducdung0909]

Dear @lminiero ,
I know you made some changes .I have downloaded newest one from https://github.com/meetecho/janus-gateway and run install.sh. As i said in previous comment, I faild on registering with newest and still successful registering with older source. The test with newest source ( updated 1-2 days ago ) give warning :

There's a message for JANUS SIP plugin
[ERR] [janus_sip.c:janus_sip_handler:678:] Missing or invalid element (proxy)

[lminiero]

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.

[ducdung0909]

Dear @lminiero ,
Sorry, I forget replace the newest client. Now I can register and make a call with clearly voice. One more thing, the caller is asked for using microphone but the callee is asked for using both camera and microphone. This is a only a voice call so we can disable camera request at callee side ?
Thank you very much.

[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.

[ducdung0909]

Hi,
Sorry for post many seperated comments, I keep posting as soon as possible. I run the command "sip show peers" in Asterisk CLI and see that.

Name/username Host
1000/1000 192.168.16.64
1001/1001 192.168.16.64
1002/1002 192.168.16.164

but

• user 1000 is on web browser at 192.168.16.194
• user 1001 is on web browser at 192.168.16.195,
• 1002 is on softphone at 192.168.16.164 ,
• and 192.168.16.64 is IP of Asterisk server, is it normal ? I stills make calls successfully both ways ( web to web, web to soft, soft to web ).

[lminiero]

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.

[ducdung0909]

Hi @lminiero ,
The request of using camera at callee dissappear now, audio call is clear.
About the host IP, that's right, I clearly understand. Thank you so much.

[lminiero]

Ok closing the issue then.