Troubleshooting WebRTC Issues

Asterisk Asterisk Support
1

Here is a little guide to troubleshoot webrtc issues with Asterisk.

Always try to use the latest WebRTC API with the latest Asterisk branch(11 or 12). Be sure you have the icessuport enabled in the rtp.conf

ISSUE: I get this response on JSSIP or SIPML5 debug:tRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd.
This issue is caused because you asterisk don’t have ICE support, you can solve that by installing the uuid/libuuid and uuid-devel/libuuid-devel packages on your system. Then recompile asterisk(be sure to rerun the configure script before the make command).

ISSUE: I get this response on JSSIP or SIPML5:Failed to get local SDP
This issue can be caused as the same cuase described above, be sure you aren’t getting the above message in the javascript debug.

If you aren’t getting the lack of ice-ufrag and ice-pwd then check your peer settings in the asterisk side. Remember you need in your peer settings:
– avpf=yes
– icesupport=yes
– encryption=yes
– transport=ws

And disable the videosupport if you don’t set the use of the vide in the WebRTC API.

ISSUE: I get this in the Asterisk cli Rejecting secure audio stream without encryption details: audio RTP/SAVPF:
Be sure that each webrtc peer have in the settings these values:
– avpf=yes
– encryption=yes

Setting that globally doesn’t work.

ISSUE: Not getting audio at all
Usually this is caused by two reasons:
** Wrong NAT settings in your Asterisk side.
** The ICE server is setting the wrong IP to use in the SDP.

Fixing the NAT settings:
First check how are you making your tests in your enviroment:
If you are in the same LAN then your webrtc peers no need NAT at all, so define the setting nat=no.

If you are connecting your peer to a NATed Asterisk box then define the setting nat=force_rpot,comedia.

Workaround for the ICE issue:
If setting the nat settings correctly doesn’t solve your issue then you need to get the Debug on Asterisk and Chrome.
On the Asterisk side you need to enable the SIP debug by running the command:

Then try a call from your WebRTC API you will get an output like this:

[code]<— SIP read from WS:[client-ip]:[client port] —>
INVITE sip:[called number]@[asterisk-server-ip] SIP/2.0

bunch of SIP HEADERS

c=IN IP4 [client-ip]
more sip headers and SDP options
[/code]

If the [Client-iP] is different in the <—SIP read from WS: ----> part and in the c= IN IP4 then the ICE server is choosing the wrong IP, for example this output:

[code]<— SIP read from WS:179.213.21.254:59385 —>
INVITE sip:1000@mypublicip.no-ip.org SIP/2.0
Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051
Max-Forwards: 69
To: sip:1000@mypublicip.no-ip.org
From: “G” sip:8001@mypublicip.no-ip.org;tag=i6i4nniqbv
Call-ID: cgu3s60etphsvr40mneg
CSeq: 5919 INVITE
Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 2242

v=0
o=- 3652154159910496246 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36
m=audio 54280 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 192.168.101.103
a=rtcp:54280 IN IP4 192.168.101.103
a=candidate:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0
a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0
a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0
a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0
a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0
a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0
a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0
a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0
a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0
a=ice-ufrag:eNWxE4gTTqnCVyMu
a=ice-pwd:ZoVKNOv2HlfHE9VNyW3NSIAO
a=ice-options:google-ice
a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:tCMlVm6sT9MJMxbzKEmsDV1e3VVvNa5MwkUKPKzN
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qHzInMoGik8HNFMVgi6IjHdClreeyslHlw11nvn8
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3951139034 cname:1GEOwdZBp2a0BbWt
a=ssrc:3951139034 msid:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 9cda3eb5-5a5c-4cb3-ad6d-0a153309d435
a=ssrc:3951139034 mslabel:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36
a=ssrc:3951139034 label:9cda3eb5-5a5c-4cb3-ad6d-0a153309d435
<------------->[/code]
Note that the signalling came with the IP 179.213.21.254:59385 meanwhile the SDP use the IP 192.168.101.103
That can be caused by a bad NAT setting config.

If you received an output like:

[code] <— SIP read from WS:192.168.0.106:51709 —>
INVITE sip:202@192.168.0.109 SIP/2.0
Via: SIP/2.0/WS hat7f7lbgktt.invalid;branch=z9hG4bK1640168
Max-Forwards: 69
To: sip:202@192.168.0.109
From: sip:200@192.168.0.109;tag=p8tsev60np
Call-ID: 6egc17kuvl2je0o87b4o
CSeq: 2789 INVITE
Authorization: Digest algorithm=MD5, username=“200”, realm=“asterisk”, nonce=“21fdd8c0”, uri="sip:202@192.168.0.109", response="077edbb83494203045c541edc89fb9e5"
Contact: sip:ugioubt2@hat7f7lbgktt.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 2150

v=0
o=- 7117681503785597723 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS kmOganHLCy2vMNaUchD012reLNNUCaUv7Dtp
m=audio 64863 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 147.175.216.240
a=rtcp:64863 IN IP4 147.175.216.240
a=candidate:1560575937 1 udp 2122260223 169.254.65.8 64862 typ host generation 0
a=candidate:1560575937 2 udp 2122260223 169.254.65.8 64862 typ host generation 0
a=candidate:1516508954 1 udp 2122194687 192.168.0.106 64863 typ host generation 0
a=candidate:1516508954 2 udp 2122194687 192.168.0.106 64863 typ host generation 0
a=candidate:327648049 1 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:327648049 2 tcp 1518280447 169.254.65.8 0 typ host generation 0
a=candidate:350743530 1 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=candidate:350743530 2 tcp 1518214911 192.168.0.106 0 typ host generation 0
a=candidate:3677098414 1 udp 1685987071 147.175.216.240 64863 typ srflx raddr 192.168.0.106 rport 64863 generation 0
a=candidate:3677098414 2 udp 1685987071 147.175.216.240 64863 typ srflx raddr 192.168.0.106 rport 64863 generation 0
a=ice-ufrag:agqGIusA/g+2fZhV
a=ice-pwd:CVzSoqyvRnAYZeqmpqyWg9Bg
a=ice-options:google-ice
a=fingerprint:sha-256 EB:69:64:1E:F8:7A:E6:9B:4C:A5:FA:16:F1:2D:5C:66:A4:64:E5:39:68:DD:E2:E3:FF:F0:1C:C8:31:7E:62:84
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:dHjkiwclb9U+ZuoKJ4+s30/i5j/nsLrWzMG3IzCF
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:J3A2MBWJgw5LgudzvuQuBm5W+zXcNk2t0ATUvlJq
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:729560918 cname:6uOjmNuEFKaaJcWh
a=ssrc:729560918 msid:kmOganHLCy2vMNaUchD012reLNNUCaUv7Dtp c7fe212f-f4b7-401c-9e9e-d47988069681
a=ssrc:729560918 mslabel:kmOganHLCy2vMNaUchD012reLNNUCaUv7Dtp
a=ssrc:729560918 label:c7fe212f-f4b7-401c-9e9e-d47988069681
<-------------> [/code]
Note that the Sip signalling come from a LAN IP while the SDP is using a Public IP address, so the ICE server is setting your public IP in the SDP to solve that set the stun as null in your API.

For SIPML5 set as :

For JsSIP set as:

That will force to use the local IP address.

ISSUE:I have all settings correct but I don’t have audio
Check that your asterisk is sending the Audio to the correct IP using ICE. Enable the RTP debug in asterisk and be sure that you see an output like this:

Got RTP packet from 10.0.1.102:41739 (type 00, seq 007702, ts 2401199386, len 000160) Sent RTP packet to 10.0.1.102:41739 (via ICE) (type 00, seq 009610, ts 031680, len 4294967284)
Please note the (Via ICE) if you dont see that maybe the asterisk don’t have the icesupport enable in the pper or in the rtp.conf file.

How to collect debugs

– For get the SIP debug in Asterisk:

– For get the RTP debug in Asterisk:

– For get the Chrome log(Javascript debug):

If you are using Doubango’s media gateway(WebRTC2SIP) then disable in the peer configuration the setting:
avpf=yes
encription=yes
icesupport=yes

ISSUE I have Installed libuuid/uuid but asterisk never send the ice-ufrag & ice-pwd in the SDP
There is a bug report for Branch 11.7 to 11.8 issues.asterisk.org/jira/browse/ASTERISK-23425 where Asterisk never send those settings Upgrade asterisk to 11.9 and it should work out of the box.

Added 4-June-2014
ISSUE After upgrade Chrome to the version 35 SIPML5 stop working
As all we know Chrome dropped the support of SDES as default and now use DTLS-SRTP, you can follow the thread in the Asterisk Jira page and see the progress with the patches.

The developers from doubango kindly give me a link to a temporary work around(until SDES will be removed in v37) code.google.com/p/sipml5/issues/detail?id=183

2

VP8 is not supported in Asterisk 11. If you patched the asterisk check if its working so far the issue is very clear in the asterisk side:

channel.c:940 ast_best_codec: Don't know any of (vp8) formats [May 27 12:20:25] NOTICE[7839][C-00000003]: chan_sip.c:29842 sip_request_call: Asked to get a channel of unsupported format (vp8) while capability is (gsm|ulaw|alaw|h263|vp8|testlaw) [May 27 12:20:25] WARNING[7839][C-00000003]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 58 - Bearer capability not available)

Since you are using webrtc2sip it will do the conversion between vp8 and h263 if you are using video capabilities. So remove conflictive codecs and try again.

3

[size=85]Malcolmd please if you can merge this with the another thread that would be great. Or lock the thread too as you prefer.
[/size]

This is for the latest Asterisk(today: Asterisk-11.11.0) without any patches or funny configurations and for the latest SIPML5 API(today: 1.4.217) from Doubango without any patches too on a CENTOS 6.

1.- Download Asterisk:

wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.11.0.tar.gz

2.- Use the install_prereq script for install almost all dependencies:

cd ${ASTERISKSOURCE_PATH}/contrib/scripts/ ./install_prereq install

3.- Just to be sure install uuid support

4.- Run the configure script and then confirm that almost all modules are enable including the res_http_websocket, the res_rtp_multicast and res_rtp_asterisk.

cd ../../ ./configure && make menuselect

5.- When finish install Asterisk:

6.- Create the Certificates(change10.0.1.108 with your IP or your DNS):

mkdir /etc/asterisk/keys cd ${ASTERISKSOURCE_PATH}/contrib/scripts/ ./ast_tls_cert -C 10.0.1.108 -O "My Super Company" -d /etc/asterisk/keys

7.- Configure your sip.conf and create some peers for your WebRTC I’m using these settings:

[code][general]

context=guest
transport=udp,ws
rtcachefriends=yes
allowguest=yes
limitonpeers=yes
callcounter=yes
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
;externhost=set_your_externhost_here
externrefresh=150
localnet=set_your_localnet_here ;i.e. 10.0.1.0/255.255.255.0
disallow=all
;allow=g729 ; First disallow all codecs
allow=gsm
allow=ulaw ; Allow codecs in order of preference
allow=alaw ; Allow codecs in order of preference
language=en ; Default language setting for all users/peers
callcounter=yes
limitonpeers=yes
callevents=yes
useragent=Digital-Merge_UA
realm=asterisk
nat=force_rport,comedia

[5005]
type=friend
secret=MySuperSecr3t4dm5005$!
host=dynamic
context=wrtc
disallow=all
allow=ulaw
allow=alaw
;allow=g729
;allow=gsm
;allow=h263p
;allow=h264
dtmf=auto
;videosupport=yes
transport=ws,udp
avpf=yes
nat=force_rport,comedia
callerid=“WebRTC”<5005>
encryption=yes
qualify=yes
icesupport=yes
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS

[/code]
Every peer need the DTLS PART! This is not a global setting.

8.- Configure the http.conf as described in the asterisk wiki, i have only:

[general] enabled=yes bindaddr=0.0.0.0 bindport=8088

9.- Configure the rtp.conf as described in the asterisk wiki, I have only:

[general] rtpstart=19000 rtpend=21000 icesupport=true stunaddr=stun.l.google.com:19302

10.- Start Asterisk.

11.- Create your SuperKiller App using SIPML5 pointing to your asterisk server like github.com/navaismo/Elastix-Age … /js/ml5.js

12.- Test and enjoy! If you have a supermachine this can be done in almost 40mins YAY!

Here pastebin.com/hJhnniBD you can take a look on a complete SIP DEBUG for a successfull call between a SIPML5 Client and a PSTN Number including the RTP DEBUG with the “VIA ICE” text and the DTLS support.

If you are using the Doubango’s WebRTC2SIP media gateway use the same certs configure as described in this topic: groups.google.com/forum/#!msg/d … bMIh5rGdYJ You can use the same certs generated by asterisk. Don’t enable the encryption setting in the peer.

This my last attempt to help the community about WebRTC and Asterisk. Please avoid send me private messages, tweets or emails READ carefully the firts part in this thread: viewtopic.php?f=1&t=90167

Almost all scenarios are described there, Why I know that? Because I have experienced those issues too.

I don’t use JsSIP because the devs are kind annoying so if you want support for JsSIP ask them and hope they don’t kick you out as me and others.

Stop complaining about the month or the week you have spent on this matter, we already have like a year or more dealing with it and your complians.

If you want ASAP response or help HIRE SOMEONE. WebRTC and Asterisk aren’t for N00bs, yes you have used linux and you can make web pages but you need more than that.

If you are wondering if I provide custom installation for you, the answer is YES this is my fee for a complete setup: $2000USD paypal in front.

And again STOP send me emails, privates and tweets this is annoying.

Finally be gratefull with the people involved in this, the people in the JIRA PAGE(issues.asterisk.org/jira/browse/ASTERISK-22961) whom make this possible and of course the Asterisk Devs like Joshua Colp, Mamadou from Doubango, etc they are the Real MVP :’( of this.

Good Luck! And READ READ and finally READ again this forum, google and the doubango discuss group.

Atte:
Navaismo
A retired Linux-Asterisk User.
[size=85]Sorry for my english[/size]

1 Like
4

hi navaismo
sorry for the distable , but i have follow your troubleshooying webrtc issues step by step,
at last step , when i making “sip set debug on” the the ip are identics,i have set stun to “null” on my api (sipjs)
i got the problem “Rejecting secure audio stream without encryption details: audio 5023 RTP/SAVPF 0 3 8”

At this this point ,i just want to nkow if it’s possible to make a conversation to a native sip fone (linphone) to client webrtc on a browser(chrome) trough asterisk server(sip server)
thanks for your availability and your help.
i’am greatful

5

Open a new thread and provide all the information requested above. That error means that you haven’t cofigured asterisk properly or using a bad branch. Do not post here.

6

Hello guys,

I have successfully configured webrtc with jssip but call has been disconnected immediately and throwing an error bad media description, i was tested in local LAN environment and necessary dependencies also installed please educate me to resolved this error.

Regards

Swamy HK

7

Open a new thread and share all the details and logs to get help.

8

Asterisk sipjs webrtc set up ---------------- Call gets connected and then gets exited instantly

Logs on Asterisk CLI

== WebSocket connection from ‘192.168.0.61:59084’ for protocol ‘sip’ accepted using version ‘13’
– Registered SIP ‘199’ at 192.168.0.61:59084
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP RTP CoS mark 5
> 0x7f7aa000c1e0 – Strict RTP learning after remote address set to: 27.106.101.236:40496
– Executing [200@default:1] Answer(“SIP/199-00000004”, “”) in new stack
== Spawn extension (default, 200, 1) exited non-zero on ‘SIP/199-00000004’

Logs on browser :::

INVITE sip:200@192.168.0.126 SIP/2.0

Via: SIP/2.0/WSS 3bcnd1986nui.invalid;branch=z9hG4bK7481270

Max-Forwards: 70

To: sip:200@192.168.0.126

From: sip:199@192.168.0.126;tag=2ibcovv4b6

Call-ID: ltrevhi8sda38ibi6m3u

CSeq: 2895 INVITE

Authorization: Digest algorithm=MD5, username=“199”, realm=“192.168.0.126”, nonce=“707e35f3”, uri="sip:200@192.168.0.126", response=“69086bf1ef534e82623b35719ed99f2b”

Contact: sip:dsn2gk6e@3bcnd1986nui.invalid;transport=ws;ob

Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

Supported: outbound

User-Agent: SIP.js/0.11.2

Content-Type: application/sdp

Content-Length: 1520

v=0

o=mozilla…THIS_IS_SDPARTA-61.0.1 3428808818377831462 0 IN IP4 0.0.0.0

s=-

t=0 0

a=sendrecv

a=fingerprint:sha-256 CE:BA:A4:E8:F9:B8:1D:2B:23:D8:69:B6:AC:F6:6F:48:01:AE:8F:D9:B4:AE:35:2F:CC:CA:49:EF:74:FE:39:B6

a=group:BUNDLE sdparta_0

a=ice-options:trickle

a=msid-semantic:WMS *

m=audio 40496 UDP/TLS/RTP/SAVPF 109 9 0 8 101

c=IN IP4 27.106.101.236

a=candidate:0 1 UDP 2122252543 192.168.0.61 40496 typ host

a=candidate:2 1 TCP 2105524479 192.168.0.61 9 typ host tcptype active

a=candidate:0 2 UDP 2122252542 192.168.0.61 36720 typ host

a=candidate:2 2 TCP 2105524478 192.168.0.61 9 typ host tcptype active

a=candidate:1 1 UDP 1686052863 27.106.101.236 40496 typ srflx raddr 192.168.0.61 rport 40496

a=candidate:1 2 UDP 1686052862 27.106.101.236 36720 typ srflx raddr 192.168.0.61 rport 36720

a=sendrecv

a=end-of-candidates

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level

a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level

a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid

a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1

a=fmtp:101 0-15

a=ice-pwd:9f0da746b20f68e1f56d5e80ef9a3f8e

a=ice-ufrag:7e1c5491

a=mid:sdparta_0

a=msid:{6b5f7ad0-24c4-4379-95ef-0ba4f32aeb1e} {47ea5a85-fca4-4ac1-a653-602e55d9e43e}

a=rtcp:36720 IN IP4 27.106.101.236

a=rtcp-mux

a=rtpmap:109 opus/48000/2

a=rtpmap:9 G722/8000/1

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=setup:actpass

a=ssrc:2116564091 cname:{c58b2fca-fe62-4557-ad0d-b23dfb16b36e}

sip-0.11.2.min.js:1:206131
Thu Aug 30 2018 12:18:45 GMT+0530 (IST) | sip.transport | received WebSocket text message:

SIP/2.0 100 Trying

Via: SIP/2.0/WSS 3bcnd1986nui.invalid;branch=z9hG4bK7481270;received=192.168.0.61

From: sip:199@192.168.0.126;tag=2ibcovv4b6

To: sip:200@192.168.0.126

Call-ID: ltrevhi8sda38ibi6m3u

CSeq: 2895 INVITE

Server: Asterisk PBX 13.20.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: sip:200@192.168.0.126:5060;transport=ws

Content-Length: 0

sip-0.11.2.min.js:1:206131
Thu Aug 30 2018 12:18:45 GMT+0530 (IST) | sip.transport | received WebSocket text message:

SIP/2.0 200 OK

Via: SIP/2.0/WSS 3bcnd1986nui.invalid;branch=z9hG4bK7481270;received=192.168.0.61

From: sip:199@192.168.0.126;tag=2ibcovv4b6

To: sip:200@192.168.0.126;tag=as47c1256c

Call-ID: ltrevhi8sda38ibi6m3u

CSeq: 2895 INVITE

Server: Asterisk PBX 13.20.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: sip:200@192.168.0.126:5060;transport=ws

Content-Type: application/sdp

Content-Length: 407

v=0

o=root 827776922 827776922 IN IP4 192.168.0.126

s=Asterisk PBX 13.20.0

c=IN IP4 192.168.0.126

t=0 0

m=audio 19196 RTP/SAVPF 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=maxptime:150

a=connection:new

a=setup:active

a=fingerprint:SHA-256 09:68:3B:41:55:7F:28:90:55:BB:F5:56:C9:0F:9B:36:AB:63:7D:A5:05:55:13:AD:80:83:FC:AB:3C:86:6F:33

a=rtcp-mux

a=sendrecv

sip-0.11.2.min.js:1:206131
Thu Aug 30 2018 12:18:45 GMT+0530 (IST) | sip.dialog | new UAC dialog created with status CONFIRMED sip-0.11.2.min.js:1:206131
DOMException: “Invalid description, no ice-ufrag attribute” sip-0.11.2.min.js:1:206131
DOMException: “Invalid description, no ice-ufrag attribute” sip-0.11.2.min.js:1:206131
Thu Aug 30 2018 12:18:45 GMT+0530 (IST) | sip.transport | sending WebSocket message:

ACK sip:200@192.168.0.126:5060;transport=ws SIP/2.0

Via: SIP/2.0/WSS 3bcnd1986nui.invalid;branch=z9hG4bK8047133

Max-Forwards: 70

To: sip:200@192.168.0.126;tag=as47c1256c

From: sip:199@192.168.0.126;tag=2ibcovv4b6

Call-ID: ltrevhi8sda38ibi6m3u

CSeq: 2895 ACK

Supported: outbound

User-Agent: SIP.js/0.11.2

Content-Length: 0

sip-0.11.2.min.js:1:206131
Thu Aug 30 2018 12:18:45 GMT+0530 (IST) | sip.transport | sending WebSocket message:

BYE sip:200@192.168.0.126:5060;transport=ws SIP/2.0

Via: SIP/2.0/WSS 3bcnd1986nui.invalid;branch=z9hG4bK7797377

Max-Forwards: 70

To: sip:200@192.168.0.126;tag=as47c1256c

From: sip:199@192.168.0.126;tag=2ibcovv4b6

Call-ID: ltrevhi8sda38ibi6m3u

CSeq: 2896 BYE

Reason: SIP;cause=488;text=“Not Acceptable Here”

Supported: outbound

User-Agent: SIP.js/0.11.2

Content-Length: 0

sip-0.11.2.min.js:1:206131
Thu Aug 30 2018 12:18:45 GMT+0530 (IST) | sip.transport | received WebSocket text message:

SIP/2.0 200 OK

Via: SIP/2.0/WSS 3bcnd1986nui.invalid;branch=z9hG4bK7797377;received=192.168.0.61

From: sip:199@192.168.0.126;tag=2ibcovv4b6

To: sip:200@192.168.0.126;tag=as47c1256c

Call-ID: ltrevhi8sda38ibi6m3u

CSeq: 2896 BYE

Server: Asterisk PBX 13.20.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

9

Asterisk has not been built with ICE support, or it is not enabled. ICE is mandatory for WebRTC.