Question on Web Call Server and Asterisk

roman

New Member
Trying to configure Flashphoner for calls via Asterisk using HTML5 websockets.
I open PhoneJS.html in Firefox and try to call, but get this error instead:
Code:
chan_sip.c process_sdp: Received AVP profile in audio offer but AVPF is enabled: audio 31002 RTP/AVP 8 0 18 100 111
chan_sip.c:process_sdp: Failing due to no acceptable offer found
 
Last edited:

Max

Administrator
Staff member
WCS does not require WebRTC support from the Asterisk-server, not does it requires any specific setup of AVPF. Web Call Server will interact with Asterisk vi standard SIP and RTP protocols and when working with a browser it uses WebRTC(SRTP/AVPF). Therefore you don't need to configure in Asterisk aside from common SIP peer configuration of course.
 
I am trying a webrtc-sip via Asterisk call with Asterisk 14 and WCS Server version FlashphonerWebCallServer-5.0.2159. I am using the demo application for WebRTC(Phone and Phone Video) and Bria for SIP End Point. I am getting the following issue in the console of Asterisk
[Apr 5 15:36:51] ERROR[61043][C-00000001]: chan_sip.c:5933 dialog_initialize_dtls_srtp: No DTLS-SRTP support present on engine for RTP instance '0x7f3614009500', was it compiled with support for it?
[Apr 5 15:36:51] NOTICE[61043][C-00000001]: chan_sip.c:26201 handle_request_invite: Failed to authenticate device "1060" <sip:1060@192.168.30.156>;tag=fce5ad16
Can anyone please help me on this since I am unable to debug and the same and I'm stuck with this since a couple of days.

This is my http.conf
[general]
enabled=yes
bindaddr=0.0.0.0 ; Replace this with your IP address
bindport=8088 ; Replace this with the port you want to listen on
;sip.conf
[general]
realm=192.168.30.156 ; Replace this with your IP address
udpbindaddr=192.168.30.156 ; Replace this with your IP address
transport=udp

[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=password ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
nat=no
canreinvite=no
qualify=no
transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
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
dtlscafile=/etc/asterisk/keys/ca.crt
allow=alaw
media_encryption=dtls

[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=password
context=default
disallow=all
allow=alaw
qualify=no
;rtp.conf
[general]
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=stun.l.google.com:19302
;extensions.conf
;extensions.conf
[default]
exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060
exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client registered to 1061
Have checked the configs in Asterisk and it seems to be fine.
Also I just wanted to know if the FlashPhoner client can support SRTP since I don't see the Offer/Answer having the fingerprint.
Getting the following error in the Asterisk debug Logs
han_sip.c:10726 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
Attached the logs for the WEB-SIP and SIP-WEB logs.
Can anyone give an insight on this?Thanks for the help in advance.Really appreciate the help.

Thanks,
Karthik
 

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Last edited:

Max

Administrator
Staff member
Hello
Web Call Server communicates with Asterisk as a legacy SIP peer and operates with browser as WebRTC peer.
Browser <-- WebRTC --> WCS <-- legacy SIP/RTP --> Asterisk PBX
So you have to configure peers for WCS as legacy SIP peers on your Asterisk server.

Please create a new topic in the Web Call Server 5 branch of forum. This branch is out of date.
 
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