How can I send tcp audio packet?

nakajima-akinori · March 26, 2019, 4:08am

I came from WebRTC to SIP. In the case of webrtc, turn tcp is possible for this purpose, but I can’t find the any solution or configuration in the sip of asterisk.
tcp is used in the strict company firewall which do not allow udp.

jcolp · March 26, 2019, 9:08am

Asterisk does not support TURN TCP.

nakajima-akinori · March 26, 2019, 12:01pm

Thanks for the reply.
But enabling the ice flag in the client side like pjsip or Zoiper can use the tcp turn server?
In this case, only turn udp can work?

nakajima-akinori · March 26, 2019, 12:02pm

I also found the option in pjsip.conf
ice_support=yes

github.com

asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L388


;transport=transport-udp
;context=from-internal
;disallow=all
;allow=ulaw
;auth=6002
;aors=6002
;direct_media=no
;rtp_symmetric=yes
;force_rport=yes
;rewrite_contact=yes  ; necessary if endpoint does not know/register public ip:port
;ice_support=yes   ;This is specific to clients that support NAT traversal
                   ;for media via ICE,STUN,TURN. See the wiki at:
                   ;https://wiki.asterisk.org/wiki/x/D4FHAQ
                   ;for a deeper explanation of this topic.


;[6002]
;type=auth
;auth_type=userpass
;password=6002
;username=6002

jcolp · March 26, 2019, 12:14pm

ICE is supported, but any ICE candidates that use TCP will not be considered and Asterisk itself can not be configured for it.

nakajima-akinori · March 27, 2019, 1:22am

HI jcolp. Thanks for the reply. I got it. Then, I should use the SIP NAT traversal without ICE in the case that no need for Chrome WebRTC Browser <=> Asterisk.