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.

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