Asterisk 18 - no RTP

Asterisk Asterisk SIP
1

I’ve just upgraded from Asterisk 16 to Aterisk 18. I can’t figure out why I have no RTP.

So all SIP signalling works fine. However for the inbound RTP, asterisk sends a request with the RTP port.

I have soft client on 10.8.0.2.
asterisk lan side = 10.8.0.1 and 192.168.1.1
external here is = x.x.x.x

100	1.154821	203.94.33.147	161.29.194.21	SIP/SDP	1030	Request: INVITE sip:0800000000@161.29.194.21:5060;transport=udp, in-dialog |

here is Asterisk saying where to send RTP:

m=audio 62413 RTP/AVP 8 101

But then Asterisk isn’t listening on that port

asterisk  485977        asterisk   31u  IPv4 2700959      0t0  TCP 192.168.1.1:5060->10.8.0.2:61081 (ESTABLISHED)
asterisk  485977        asterisk   32u  IPv6 2701781      0t0  UDP *:11740
asterisk  485977        asterisk   33u  IPv4 2702751      0t0  UDP x.x.x.x:10285
asterisk  485977        asterisk   38u  IPv6 2701784      0t0  UDP *:10284
asterisk  485977        asterisk   39u  IPv4 2699970      0t0  UDP 10.8.0.1:11741

Then my server responds with port unreachable

Why isn’t asterisk listening on RTP port?

Here is pjsip config

[333]
type = endpoint
context = from-internal
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
disallow = all
allow = alaw
aors = 333
auth = auth_333

[transport-udp]
type=transport
protocol=udp ;udp,tcp,tls,ws,wss
bind=0.0.0.0
local_net=192.168.1.0/24
local_net=10.8.0.0/24
external_media_address=x.x.x.x
external_signaling_address=x.x.x.x
2

You’d need to provide an actual full SIP trace, along with the output of “rtp set debug on”.

3

Thanks, I figured it out, for some reason it was trying to do direct media. I thought for trunks, direct_media was defaulted to no. Just added direct_media=no to the endpoints and it all started working

4

There is no inherent concept of “trunks”. Defaults apply equally to all.

5

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