[WEBRTC]Asterisk12:Called with SDP without ice-ufrag ice-pwd

Asterisk Asterisk Support
1

Hello,

I’m trying to use Sip.js with an Asterisk 12 installation. The call is done correctly but when answered by my sip phone, Sip.js end directly the call. The error message is:
sip.inviteclientcontext | Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd.

uuid and uuid-dev are installed. The problem come from the fact that Asterisk don’t use ICE. Can someone help me with that issue? The asterisk’s response:

[size=85]SIP/2.0 200 OK
Via: SIP/2.0/WS 192.0.2.148;branch=z9hG4bK3038236;received=127.0.0.1
From: sip:1060@127.0.0.1;tag=sm89j6j6gm
To: sip:1061@127.0.0.1;tag=as1b4ff48e
Call-ID: 1mrda29lqq8d1l6dmjos
CSeq: 2869 INVITE
Server: Asterisk PBX 12.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1061@127.0.0.1:5060;transport=WS
Content-Type: application/sdp
Content-Length: 468

v=0
o=root 1182539646 1182539646 IN IP4 127.0.0.1
s=Asterisk PBX 12.6.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 13232 RTP/SAVPF 0 8 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:150
a=connection:new
a=setup:active
a=fingerprint:SHA-256 97:5F:D2:63:A1:BA:5B:C0:47:A3:61:F7:0D:76:81:42:8F:BB:48:7B:28:01:B0:57:77:B4:4B:2E:24:1B:E6:87
a=sendrecv
m=video 0 UDP/TLS/RTP/SAVPF 100 116 117 96
[/size]

My sip.conf file contains ‘icesupport=yes’ globally and 2 clients:

[size=85][1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=1234 ; 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
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

[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=1234
context=default[/size]

2

Check your rtp.conf for the icesupport too.

3

The line “icesupport = true” is present in the rtp.conf

4

What is the output of: rpm -qa | grep uuid && echo && ldd /usr/lib/asterisk/modules/res_rtp_asterisk.so && echo && ls -lha /lib/libuu* && cat /etc/issue

replace the rpm and paths if you are using other OS than CentOS

5

I have this setup working successfully in Asterisk 11. Although your version is 12, I though it may be related.
If I install iksemel library and enable motif,jingle or gtalk I get the exact behavior you described in my SipMl5 client.

I have a very very simple Asterisk configuration which I can share:
github.com/spicyramen/llamato/t … iaServices

  1. Install Asterisk 11.13.1 with srtp
  2. create sample configs (make samples)
  3. Create keys with ast_tls_cert for DTLS
  4. overwrite files with above files
  5. Start Asterisk

Btw navaismo that command doesnt return anything in my Ubuntu 14.04 AWS.

6

[quote=“navaismo”]

replace the rpm and paths if you are using other OS than CentOS[/quote]

7

Hi,

I face the same issue.

rpm -qa | grep uuid && echo && ldd /usr/lib/asterisk/modules/res_rtp_asterisk.so && echo && ls -lha /lib/libuu* && cat /etc/issue returns nothing for me.

Can you please advice?

Thank you in advance

Christodoulos

8

Are you using CentOS? If not then replace with appropiate commands of your distro.

9

Hi,

I’m facing the same issue as christod on Ubuntu 14.04/Asterisk 11.16.0.

Any suggestions?

Cheers

10

Hi,

The output to ldd command for my distribution is the following:

#cd /usr/lib/asterisk/modules/
root@debian:/usr/lib/asterisk/modules# ldd res_rtp_asterisk.so
linux-vdso.so.1 (0x00007fff1f5ff000)
libuuid.so.1 => /lib64/libuuid.so.1 (0x00007ff186e9a000)
libpthread.so.0 => /lib64/libpthread.so.0 (0x00007ff186c7d000)
libc.so.6 => /lib64/libc.so.6 (0x00007ff1868b2000)
/lib64/ld-linux-x86-64.so.2 (0x00007ff1872d4000)

Can you please advice why ICE support is not passed when calling out and get the following error in the js debug file “Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd”

Thank you

Chris

11

Hello,

I’ve got the similar problem with one of my asterisk 11.21.2 - no “ice” tags in SDP.

I’m using Debian 7, with latest available stable packages installed, build asterisk from the sources with ./install-prereq install.

Rtp and sip configs contains all similar lines to enable icesupport, but SIP SDP still not contains it!

asterisk debug say no reason why it so

What should i check next?
How could i debug more?

Any suggestions?
Thanks