*RESOLVED*WebRTC Chome - caller's side says "Remote Ringing"

Asterisk Asterisk Support
1

Hello All,

If someone can point me in the right direction that would be greatly appreciated!

The issue: after the call is accepted, the caller’s side is not notified and still says “Remote Ringing” Eventually the call drops. The callee’s side says “In Call” and never ends until manually stopped.

I’ve collected the logs and my configuration files:

Config files removed

js_console.log: pastebin.com/1hYcjpXw

sip_debug.log: pastebin.com/eDTUZNKn

Thanks!

2

With a quick look:

In the SDP the WS peer has in the ‘C’ field this IP address–>c=IN IP4 192.168.56.1 but the INVITE came from 10.168.107.7

If you are on NAT environment check your peer’s nat settings.

Second thing, you don’t have the AVPF setting enabled in your peer configuration and also the ICE support.

Hint: Sanitize your configs files removing all comment section.

3

Thanks navaismo,

After adding the below to my sip.conf users, the issue is resolved.

avpf = yes
icesupport = yes
videosupport=no
directmedia=no

Thanks again for the quick and helpful replies.

4

I have a follow up question, if that is OK.

I am unable to make a call from a softphone such as X-Lite to Chrome with a user that has the “encryption=yes” setting enabled.

IE: user Jerry is logged into X-Lite and tries to call user Tim logged into Chrome. If Jerry has “encryption=true” then I get the error “Failed to establish call” and it then disconnects.

Is there a way around this?

sip debug log

[quote]e[0K
<— SIP read from UDP:10.168.107.7:25166 —>

<------------->

e[KLaneAsterix*CLI>
e[0K
<— SIP read from UDP:10.168.107.7:25166 —>
INVITE sip:2002@10.168.100.160 SIP/2.0
Via: SIP/2.0/UDP 10.168.107.7:25166;branch=z9hG4bK-d8754z-e0d3b7041999855a-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:jerry@10.168.107.7:25166
To: sip:2002@10.168.100.160
From: "jerry"sip:jerry@10.168.100.160;tag=aedc813e
Call-ID: ODdiMWVmYjMzYmE3Mzg3MzE0NmZiMTI3Y2FlZDQyYTI
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp

e[KLaneAsterix*CLI>
e[0KSupported: replaces
User-Agent: X-Lite release 4.5.5 stamp 71236
Content-Length: 304

v=0
o=- 13040687825776382 1 IN IP4 10.168.107.7
s=X-Lite 4 release 4.5.5 stamp 71236
c=IN IP4 10.168.107.7
t=0 0
m=audio 54128 RTP/AVP 125 9 0 8 100 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 12 lines) —
Sending to 10.168.107.7:25166 (no NAT)
Sending to 10.168.107.7:25166 (no NAT)
Using INVITE request as basis request - ODdiMWVmYjMzYmE3Mzg3MzE0NmZiMTI3Y2FlZDQyYTI
Found peer ‘jerry’ for ‘jerry’ from 10.168.107.7:25166

<— Reliably Transmitting (no NAT) to 10.168.107.7:25166 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.168.107.7:25166;branch=z9hG4bK-d8754z-e0d3b7041999855a-1—d8754z-;received=10.168.107.7;rport=25166
From: "jerry"sip:jerry@10.168.100.160;tag=aedc813e
To: sip:2002@10.168.100.160;tag=as52e02363
Call-ID: ODdiMWVmYjMzYmE3Mzg3MzE0NmZiMTI3Y2FlZDQyYTI
CSeq: 1 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“10.168.100.160”, nonce="54024b71"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ODdiMWVmYjMzYmE3Mzg3MzE0NmZiMTI3Y2FlZDQyYTI’ in 32000 ms (Method: INVITE)

e[KLaneAsterix*CLI>
e[0K
<— SIP read from UDP:10.168.107.7:25166 —>
ACK sip:2002@10.168.100.160 SIP/2.0
Via: SIP/2.0/UDP 10.168.107.7:25166;branch=z9hG4bK-d8754z-e0d3b7041999855a-1—d8754z-;rport
Max-Forwards: 70
To: sip:2002@10.168.100.160;tag=as52e02363
From: "jerry"sip:jerry@10.168.100.160;tag=aedc813e
Call-ID: ODdiMWVmYjMzYmE3Mzg3MzE0NmZiMTI3Y2FlZDQyYTI
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

e[KLaneAsterix*CLI>
e[0K
<— SIP read from UDP:10.168.107.7:25166 —>
INVITE sip:2002@10.168.100.160 SIP/2.0
Via: SIP/2.0/UDP 10.168.107.7:25166;branch=z9hG4bK-d8754z-d53ced16ced6290d-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:jerry@10.168.107.7:25166
To: sip:2002@10.168.100.160
From: "jerry"sip:jerry@10.168.100.160;tag=aedc813e
Call-ID: ODdiMWVmYjMzYmE3Mzg3MzE0NmZiMTI3Y2FlZDQyYTI
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp

e[KLaneAsterix*CLI>
e[0KSupported: replaces
User-Agent: X-Lite release 4.5.5 stamp 71236
Authorization: Digest username=“jerry”,realm=“10.168.100.160”,nonce=“54024b71”,uri="sip:2002@10.168.100.160",response=“aaa0da170596577fe3011c1b634558a1”,algorithm=MD5
Content-Length: 304

v=0
o=- 13040687825776382 1 IN IP4 10.168.107.7
s=X-Lite 4 release 4.5.5 stamp 71236
c=IN IP4 10.168.107.7
t=0 0
m=audio 54128 RTP/AVP 125 9 0 8 100 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 12 lines) —
Sending to 10.168.107.7:25166 (no NAT)
Using INVITE request as basis request - ODdiMWVmYjMzYmE3Mzg3MzE0NmZiMTI3Y2FlZDQyYTI
Found peer ‘jerry’ for ‘jerry’ from 10.168.107.7:25166
== Using SIP RTP CoS mark 5
[Mar 30 17:16:07] e[1;33mNOTICEe[0m[10886][C-00000006]: e[1;37mchan_sip.ce[0m:e[1;37m10152e[0m e[1;37mprocess_sdpe[0m: Received AVP profile in audio answer but AVPF is enabled, disabling: audio 54128 RTP/AVP 125 9 0 8 100 101
Found RTP audio format 125
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found unknown media description format opus for ID 125
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
[Mar 30 17:16:07] e[1;31mWARNINGe[0m[10886][C-00000006]: e[1;37mchan_sip.ce[0m:e[1;37m10536e[0m e[1;37mprocess_sdpe[0m: Matched device setup to use SRTP, but request was not!

<— Reliably Transmitting (no NAT) to 10.168.107.7:25166 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.168.107.7:25166;branch=z9hG4bK-d8754z-d53ced16ced6290d-1—d8754z-;received=10.168.107.7;rport=25166
From: "jerry"sip:jerry@10.168.100.160;tag=aedc813e
To: sip:2002@10.168.100.160;tag=as52e02363
Call-ID: ODdiMWVmYjMzYmE3Mzg3MzE0NmZiMTI3Y2FlZDQyYTI
CSeq: 2 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ODdiMWVmYjMzYmE3Mzg3MzE0NmZiMTI3Y2FlZDQyYTI’ in 32000 ms (Method: INVITE)

e[KLaneAsterix*CLI>
e[0K
<— SIP read from UDP:10.168.107.7:25166 —>
ACK sip:2002@10.168.100.160 SIP/2.0
Via: SIP/2.0/UDP 10.168.107.7:25166;branch=z9hG4bK-d8754z-d53ced16ced6290d-1—d8754z-;rport
Max-Forwards: 70
To: sip:2002@10.168.100.160;tag=as52e02363
From: "jerry"sip:jerry@10.168.100.160;tag=aedc813e
Call-ID: ODdiMWVmYjMzYmE3Mzg3MzE0NmZiMTI3Y2FlZDQyYTI
CSeq: 2 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

e[KLaneAsterix*CLI>
e[0K
<— SIP read from UDP:10.168.107.7:25166 —>
SUBSCRIBE sip:2003@10.168.100.160 SIP/2.0
Via: SIP/2.0/UDP 10.168.107.7:25166;branch=z9hG4bK-d8754z-d37d7c23be7b324c-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:jerry@10.168.107.7:25166
To: sip:2003@10.168.100.160
From: "jerry"sip:jerry@10.168.100.160;tag=8c47234b
Call-ID: ZTM2NjE5ZDI5Mzg5ZTcwZjcyY2FmOGYxN2U5YzZhMTA
CSeq: 1 SUBSCRIBE
Expires: 3600
Accept: multipart/related, application/rlmi+xml, application/pidf+xml
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: eventlist
User-Agent: X-Lite release 4.5.5 stamp 71236
Event: presence
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 10.168.107.7:25166 (no NAT)
Creating new subscription
Sending to 10.168.107.7:25166 (no NAT)
list_route: hop: sip:jerry@10.168.107.7:25166
Found peer ‘jerry’ for ‘jerry’ from 10.168.107.7:25166

<— Transmitting (no NAT) to 10.168.107.7:25166 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.168.107.7:25166;branch=z9hG4bK-d8754z-d37d7c23be7b324c-1—d8754z-;received=10.168.107.7;rport=25166
From: "jerry"sip:jerry@10.168.100.160;tag=8c47234b
To: sip:2003@10.168.100.160;tag=as24dc6c25
Call-ID: ZTM2NjE5ZDI5Mzg5ZTcwZjcyY2FmOGYxN2U5YzZhMTA
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“10.168.100.160”, nonce="64656619"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZTM2NjE5ZDI5Mzg5ZTcwZjcyY2FmOGYxN2U5YzZhMTA’ in 32000 ms (Method: SUBSCRIBE)

e[KLaneAsterix*CLI>
e[0K
<— SIP read from UDP:10.168.107.7:25166 —>
SUBSCRIBE sip:2003@10.168.100.160 SIP/2.0
Via: SIP/2.0/UDP 10.168.107.7:25166;branch=z9hG4bK-d8754z-021d6458f2695348-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:jerry@10.168.107.7:25166
To: sip:2003@10.168.100.160
From: “jerry"sip:jerry@10.168.100.160;tag=8c47234b
Call-ID: ZTM2NjE5ZDI5Mzg5ZTcwZjcyY2FmOGYxN2U5YzZhMTA
CSeq: 2 SUBSCRIBE
Expires: 3600
Accept: multipart/related, application/rlmi+xml, application/pidf+xml
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: eventlist
User-Agent: X-Lite release 4.5.5 stamp 71236
Authorization: Digest username=“jerry”,realm=“10.168.100.160”,nonce=“64656619”,uri="sip:2003@10.168.100.160”,response=“bf6fc2ddc7244453977e5609c644e03e”,algorithm=MD5
Event: presence
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Creating new subscription
Sending to 10.168.107.7:25166 (no NAT)
Found peer ‘jerry’ for ‘jerry’ from 10.168.107.7:25166
Looking for 2003 in users (domain 10.168.100.160)

<— Transmitting (no NAT) to 10.168.107.7:25166 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.168.107.7:25166;branch=z9hG4bK-d8754z-021d6458f2695348-1—d8754z-;received=10.168.107.7;rport=25166
From: "jerry"sip:jerry@10.168.100.160;tag=8c47234b
To: sip:2003@10.168.100.160;tag=as24dc6c25
Call-ID: ZTM2NjE5ZDI5Mzg5ZTcwZjcyY2FmOGYxN2U5YzZhMTA
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0[/quote]

many thanks!

5

Encryption=yes for normal users only is valid if you are using TLS with SIP or for WS peers. If your normal peers(SIP softphones or SIP desk phones) doesn’t support SRTP and TLS then disable the encryption setting.

In the other hand avpf and icesupport settings are only for ws peers.

6

So is it possible to have a user that can use WebRTC along with X-lite simultaneously?

7

Only if you use the webrtc2sip media gateway(from Doubango too). Otherwise you need to create differents peers.

8

Thanks I do not have the gateway installed.

I will use separate logins for X-Lite and Chrome users.