The SIP part (sip-connector vs Asterisk connection) works well so far, the communication starts but with no audio.
I noticed that the RTP flux is sent to localhost instead of my server address (set as remote in sip-connector.cfg) and I was wondering if there is any possibility to send the RTP flow to an address which is not localhost ?
In a configuration where all the Osmocom servers (MSC, MGW, BSC…) and Asterisk are on the same machine, it got a message from my asterisk server, saying that no codec can be found to start a communication. By default, the wiki/manuals states that gsm has to be used but perhaps I am missing something in the BSC configuration, especially in the codec choice.
<--- SIP read from UDP:10.184.10.162:5069 --->
Via: SIP/2.0/UDP 10.184.10.162:5069;rport;branch=z9hG4bKpe1UXZU1amUja
Max-Forwards: 70
Call-ID: cd65c5a0-fdbf-1238-51a9-000c29cfd753
CSeq: 949096397 INVITE
User-Agent: sofia-sip/1.12.11devel
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Length: 133
v=0
o=Osmocom 0 0 IN IP4 127.0.0.1
s=GSM Call
c=IN IP4 127.0.0.1
t=0 0
m=audio 4016 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=sendrecv
<------------->
--- (13 headers 8 lines) ---
Sending to 10.184.10.162:5069 (no NAT)
Sending to 10.184.10.162:5069 (no NAT)
Using INVITE request as basis request - cd65c5a0-fdbf-1238-51a9-000c29cfd753
No matching peer for '422' from '10.184.10.162:5069'
== Using SIP RTP CoS mark 5
Got SDP version 0 and unique parts [Osmocom 0 IN IP4 127.0.0.1]
Found RTP audio format 3
Found audio description format GSM for ID 3
[2020-04-20 17:38:19] NOTICE[14918][C-00000015]: chan_sip.c:10957 process_sdp: No compatible codecs, not accepting this offer!