[asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

[Alex Villací­s Lasso]

I am having trouble getting Google Chrome to accept a WebRTC call coming from 
Asterisk, even though Firefox can (now) accept the same call without issue.

My setup is as follows:

Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around 
https://issues.asterisk.org/jira/browse/ASTERISK-25659
openssl-1.0.1e-51.el7_2.2.x86_64
[root@elx4 ~]# openssl version
OpenSSL 1.0.1e-fips 11 Feb 2013
[root@elx4 ~]# openssl ecparam -list_curves
  secp384r1 : NIST/SECG curve over a 384 bit prime field
  secp521r1 : NIST/SECG curve over a 521 bit prime field
  prime256v1: X9.62/SECG curve over a 256 bit prime field

Client:
Fedora 23 x86_64
Linphone (linphone-3.6.1-10.fc23.x86_64)
Firefox 43 (firefox-43.0.3-1.fc23.x86_64)
Google Chrome (google-chrome-stable-47.0.2526.111-1.x86_64)
SIP.js 0.7.2

I set up my SIP configuration to have two SIP accounts. Account 1000 is the 
Linphone and 1001 is the webrtc:

[general]
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.20.0)
disallow=all
allow=g723
allow=ulaw
allow=gsm
allow=alaw
allow=g729
allow=speex
allow=g722
allow=h264
allow=h263p
allow=h263
allow=h261
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscipher=ALL
tlsclientmethod=tlsv1
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
callevents=no
jbenable=no
videosupport=yes
allowguest=no
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes

[1000]
deny=0.0.0.0/0.0.0.0
secret=6ff108122cce3b0b45e0abf374c14ef4
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
icesupport=no
dtlsenable=no
dtlsverify=no
dtlssetup=actpass
encryption=no
callgroup=
pickupgroup=
dial=SIP/1000
mailbox=1000@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario 1 elx4 <1000>
callcounter=yes
faxdetect=no

[1001]
deny=0.0.0.0/0.0.0.0
secret=ce93963b0751ed9a88ec1badbc073fce
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=wss,ws,udp,tcp,tls
avpf=yes
icesupport=yes
dtlsenable=yes
dtlsverify=no
dtlssetup=actpass
dtlscertfile=/var/lib/asterisk/keys/localhost.crt
dtlsprivatekey=/var/lib/asterisk/keys/localhost.key
encryption=yes
callgroup=
pickupgroup=
dial=SIP/1001
mailbox=1001@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario Alex <1001>
callcounter=yes
faxdetect=no


With this setup, I can make calls using the SIP softphone as usual, both into and out of asterisk. After approving the certificate exceptions, I can also use the webrtc account to generate a call from either Firefox or Google Chrome into asterisk, out to 
the SIP softphone.


The problem arises when I try to make asterisk send a call into the browser. When using Firefox 43 I can receive the call normally (this required patching around ASTERISK-25659) and all is well. However, in Google Chrome, the call is rejected with a 
message of "Failed to set remote video description send parameters.." as shown in this SIP trace in the browser console:




Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | received WebSocket 
text message:

INVITE sip:8dgpkoa2@192.0.2.210;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK4f80b96d;rport
Max-Forwards: 70
From: "Anonymous" ;tag=as37a33245
To: 
Contact: 
Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Wed, 20 Jan 2016 18:54:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1799

v=0
o=root 469858785 469858785 IN IP4 10.1.0.4
s=Asterisk PBX 11.21.0
c=IN IP4 10.1.0.4
b=CT:384
t=0 0
m=audio 14814 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0e746cd50c88ce6e383ff3882acebb80
a=ice-pwd:1a9a09862254ae253f06a0bb184fd1b5
a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 14814 typ host
a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 14814 typ host
a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 14815 typ host
a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 14815 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 

Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

[Alex Villací­s Lasso]

El 20/01/16 a las 16:25, Alex Villací­s Lasso escribió:

I am having trouble getting Google Chrome to accept a WebRTC call coming from 
Asterisk, even though Firefox can (now) accept the same call without issue.

My setup is as follows:

Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around 
https://issues.asterisk.org/jira/browse/ASTERISK-25659
openssl-1.0.1e-51.el7_2.2.x86_64
[root@elx4 ~]# openssl version
OpenSSL 1.0.1e-fips 11 Feb 2013
[root@elx4 ~]# openssl ecparam -list_curves
  secp384r1 : NIST/SECG curve over a 384 bit prime field
  secp521r1 : NIST/SECG curve over a 521 bit prime field
  prime256v1: X9.62/SECG curve over a 256 bit prime field

Client:
Fedora 23 x86_64
Linphone (linphone-3.6.1-10.fc23.x86_64)
Firefox 43 (firefox-43.0.3-1.fc23.x86_64)
Google Chrome (google-chrome-stable-47.0.2526.111-1.x86_64)
SIP.js 0.7.2

I set up my SIP configuration to have two SIP accounts. Account 1000 is the 
Linphone and 1001 is the webrtc:

[general]
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.20.0)
disallow=all
allow=g723
allow=ulaw
allow=gsm
allow=alaw
allow=g729
allow=speex
allow=g722
allow=h264
allow=h263p
allow=h263
allow=h261
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscipher=ALL
tlsclientmethod=tlsv1
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
callevents=no
jbenable=no
videosupport=yes
allowguest=no
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes

[1000]
deny=0.0.0.0/0.0.0.0
secret=6ff108122cce3b0b45e0abf374c14ef4
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
icesupport=no
dtlsenable=no
dtlsverify=no
dtlssetup=actpass
encryption=no
callgroup=
pickupgroup=
dial=SIP/1000
mailbox=1000@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario 1 elx4 <1000>
callcounter=yes
faxdetect=no

[1001]
deny=0.0.0.0/0.0.0.0
secret=ce93963b0751ed9a88ec1badbc073fce
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=wss,ws,udp,tcp,tls
avpf=yes
icesupport=yes
dtlsenable=yes
dtlsverify=no
dtlssetup=actpass
dtlscertfile=/var/lib/asterisk/keys/localhost.crt
dtlsprivatekey=/var/lib/asterisk/keys/localhost.key
encryption=yes
callgroup=
pickupgroup=
dial=SIP/1001
mailbox=1001@device
permit=0.0.0.0/0.0.0.0
callerid=Usuario Alex <1001>
callcounter=yes
faxdetect=no


With this setup, I can make calls using the SIP softphone as usual, both into and out of asterisk. After approving the certificate exceptions, I can also use the webrtc account to generate a call from either Firefox or Google Chrome into asterisk, out to 
the SIP softphone.


The problem arises when I try to make asterisk send a call into the browser. When using Firefox 43 I can receive the call normally (this required patching around ASTERISK-25659) and all is well. However, in Google Chrome, the call is rejected with a 
message of "Failed to set remote video description send parameters.." as shown in this SIP trace in the browser console:




Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | received WebSocket 
text message:

INVITE sip:8dgpkoa2@192.0.2.210;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK4f80b96d;rport
Max-Forwards: 70
From: "Anonymous" ;tag=as37a33245
To: 
Contact: 
Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Wed, 20 Jan 2016 18:54:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1799

v=0
o=root 469858785 469858785 IN IP4 10.1.0.4
s=Asterisk PBX 11.21.0
c=IN IP4 10.1.0.4
b=CT:384
t=0 0
m=audio 14814 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0e746cd50c88ce6e383ff3882acebb80
a=ice-pwd:1a9a09862254ae253f06a0bb184fd1b5
a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 14814 typ host
a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 14814 typ host
a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 14815 typ host
a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 14815 typ host
a=connection:new
a=setup:actpass

Re: [asterisk-users] Incoming webrtc call succeeds in Firefox but fails in Google Chrome

[Alex Villací­s Lasso]

El 20/01/16 a las 18:33, Alex Villací­s Lasso escribió:

El 20/01/16 a las 16:25, Alex Villací­s Lasso escribió:

Partial fix: Google Chrome accepts the call if videosupport is set to "no". 
This is the SDP of the successful INVITE that Chrome accepts:

INVITE sip:8cj802p8@192.0.2.240;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK65071dc5;rport
Max-Forwards: 70
From: "Anonymous" ;tag=as474012b5
To: 
Contact: 
Call-ID: 73b82a5b6fbaab50741cd99424b1f31a@10.1.0.4:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Wed, 20 Jan 2016 23:27:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 937

v=0
o=root 2094440180 2094440180 IN IP4 10.1.0.4
s=Asterisk PBX 11.21.0
c=IN IP4 10.1.0.4
t=0 0
m=audio 18758 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:0033e2a20fe4becd1c34b13f5efcf1e3
a=ice-pwd:65693b30588f061710baa3584253eaba
a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 18758 typ host
a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 18758 typ host
a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 18759 typ host
a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 18759 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 
A1:9E:D0:11:26:AA:30:00:AF:06:87:9C:A7:C2:70:4F:A3:3F:89:B5:7D:5C:FA:90:89:7E:D0:8A:F0:72:F5:2A
a=sendrecv

However, I want to enable full video passthrough. Is this some kind of video 
codec incompatibility?



If I enable allow=vp8 to the set of allowed codecs, Chrome accepts the video 
call, but now I get no sound with the demo-congrats command.

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