On Tue, Aug 18, 2015 at 3:12 AM, Chirag Desai wrote:
> Hi all,
>
> I'm using Kamailio + Asterisk 13 (PJSIP), where Kamailio (using rtpengine)
> acts as the registrar and forwards all calls to Asterisk.
>
> This works fine when using udp / tcp and RTP. When switching to TLS/SRTP,
> the call is set up correctly, however, I get no audio.
>
> When I skip kamailio and connect my two endpoints to asterisk directly I
> get a perfect call with SRTP.
>
> The same is also true when I skip asterisk and have the call handled by
> Kamailio (using RTPEngine).
>
> In PJSIP my transports look like this:
>
> [transport-tcp]
> type=transport
> protocol=tcp;udp,tcp,tls,ws,wss
> bind=0.0.0.0:5060
> local_net=[asterisk local ip]/17
> external_media_address=[asterisk external ip]
> external_signaling_address=[asterisk external ip]
>
> [transport-tls]
> type=transport
> protocol=tls
> bind=0.0.0.0:5063
> ca_list_file=/etc/asterisk/certificates/cert.crt
> cert_file=/etc/asterisk/certificates/certificate.crt
> priv_key_file=/etc/asterisk/certificates/key.key
> method=tlsv1
>
>
> My endpoint looks like this:
>
> [kamailio]
> type=endpoint
> context=kam_out
> disallow=all
> allow=alaw
> allow=g722
> allow=ulaw
> allow=gsm
> aors=kamailio
> direct_media=no
> media_encryption=sdes
> media_address=[Asterisk Local IP]
> rtp_symmetric=yes
> force_rport=no
> rewrite_contact=yes
> outbound_proxy=sip:[Kamailio Local IP]:5060\;transport=tcp\;lr
>
> [kamailio]
> type=identify
> endpoint=kamailio
> match=[Kamailio Local IP]/17
>
> [kamailio]
> type=aor
> contact=sip:[Kamailio Local IP]:5060\;transport=tcp
>
>
> My dialplan looks like this
>
> [kam_out]
>
> exten => 1001,1,Playback(demo-echotest) ; Let them know what's going on
> same => n,Echo ; Do the echo test
> same => n,Playback(demo-echodone) ; Let them know it's over
> same => n,Hangup()
>
>
> exten => _kb-.,1,NoOp(Calling a registred user with number ${EXTEN})
> same => n,Set(callee=${PJSIP_HEADER(read,To)})
> same => n,Set(callee=${callee:5})
> same => n,Set(callee=${callee:0:-1}) ; removes the >
> same => n,Dial(PJSIP/kamailio/sip:${callee})
> same => n,Hangup()
>
> When a call comes via kamailio it comes with a prefix of 'kb' if the value
> is an extension e.g. 1000 - 1999. Otherwise users can dial a prefix of 45
> e.g. 451001 to hit the Echo Test.
>
> As mentioned the echo test works fine, however the actual call between two
> endpoints has no audio. RTP debug shows nothing. PJSIP shows two channels
> in a simple bridge, but no sound. Usually PJSIP says RTP Probation passed
> and shows the IP address but in this case it does not.
>
>
The PJSIP stack only provides SIP signalling; it doesn't interfere with the
media handling in Asterisk. The handling of media is done by the RTP engine
implementation, res_rtp_asterisk.
I don't think this is a problem, however, with res_rtp_asterisk or
Asterisk. If RTP debug doesn't show any traffic, then Asterisk is almost
certainly not receiving any media.
What does a PCAP show? I'd look at where the RTPEngine is forwarding your
RTP packets off to, and see if they are getting sent somewhere other than
Asterisk.
> I'm guessing the issue is something funny in PJSIP, although I'm not 100%
> since it does work when I turn SRTP and TLS off.
>
> For testing I'm using CsipSimple and a Snom 760. Both are set with SRTP
> mandatory and are using TLS to talk to Kamailio.
>
> When kamailio talks to asterisk it uses TCP over a local network.
>
> I've been pulling my hair out for days. I really would appreciate any
> ideas or some pointing in the right direction here.
>
> Thanks in advance,
>
> C
>
> --
> _
> -- 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
>
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_
-- 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
