VoIP Mailing List Archives

[a_villacis at palosant...]

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" <sip:anonymous@anonymous.invalid>;tag=as37a33245
To: <sip:8dgpkoa2@192.0.2.210;transport=wss>
Contact: <sip:anonymous@10.1.0.4:5060;transport=WS>
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 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
m=video 13042 UDP/TLS/RTP/SAVPF 99 98 34 31
a=ice-ufrag:68dab2b617fb033e5e3ed2821c99488a
a=ice-pwd:0ab624a87de92ca424db5af41da0cef3
a=candidate:Ha010004 1 UDP 2130706431 10.1.0.4 13042 typ host
a=candidate:Hc0a80592 1 UDP 2130706431 192.168.5.146 13042 typ host
a=candidate:Ha010004 2 UDP 2130706430 10.1.0.4 13043 typ host
a=candidate:Hc0a80592 2 UDP 2130706430 192.168.5.146 13043 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=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=rtpmap:98 H263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:31 H261/90000
a=sendrecv


sip-0.7.2.js:2892 Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.sipmessage | header "Content-Disposition" not present
sip-0.7.2.js:2892 Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | sending WebSocket message:

SIP/2.0 100 Trying
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK4f80b96d;rport
To: <sip:8dgpkoa2@192.0.2.210;transport=wss>
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37a33245
Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060
CSeq: 102 INVITE
Supported: outbound
User-Agent: SIP.js/0.7.2
Content-Length: 0



sip-0.7.2.js:2892 Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.dialog | 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060kqug9lojk8as37a33245 | new UAS dialog created with status EARLY
sip-0.7.2.js:2892 Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.inviteservercontext | 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060as37a33245 | invalid SDP (A)
sip-0.7.2.js:2892 Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.inviteservercontext | 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060as37a33245 | Failed to set remote offer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set
remote video description send parameters..
sip-0.7.2.js:2892 Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | sending WebSocket message:

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WS 10.1.0.4:5060;branch=z9hG4bK4f80b96d;rport
To: <sip:8dgpkoa2@192.0.2.210;transport=wss>;tag=kqug9lojk8
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37a33245
Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060
CSeq: 102 INVITE
Supported: outbound
User-Agent: SIP.js/0.7.2
Content-Length: 0



sip-0.7.2.js:2892 Wed Jan 20 2016 13:54:53 GMT-0500 (ECT) | sip.transport | received WebSocket text message:

ACK 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" <sip:anonymous@anonymous.invalid>;tag=as37a33245
To: <sip:8dgpkoa2@192.0.2.210;transport=wss>;tag=kqug9lojk8
Contact: <sip:anonymous@10.1.0.4:5060;transport=WS>
Call-ID: 61c6be5b44d587c80b56f98e756037ab@10.1.0.4:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.20.0)
Content-Length: 0


From what I could trace, the failing call is one on RTCPeerConnection.setRemoteDescription() with a string parameter containing the SDP body. However, the error message tells me nothing about what exactly is wrong with the SDP. A Google search with the
error message returns me only three results, which only point right into the guts of the source code of public repositories for Chrome.

I have looked into http://forums.digium.com/viewtopic.php?f=1&t=90167&sid=66fdf8cc4be5d955ba584e989a23442f as suggested by the SIP.js guide in Asterisk configuration, and none of the scenarios outlined match what I am seeing. I have tried setting
force_avp=yes without it making any difference.

Is there something that stands out as being incorrect about my configuration, or the SDP, that could explain the Chrome failure? Is there a known way to make the browser show more information, short of recompiling, so that I can get a clue on how to debug
this?

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[a_villacis at palosant...]

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

[a_villacis at palosant...]

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