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[djchillerz at gmail.com]

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

[mjordan at digium.com]

On Tue, Aug 18, 2015 at 3:12 AM, Chirag Desai <djchillerz@gmail.com (djchillerz@gmail.com)> wrote: