Hi all,
I am new to the use of the Osmocom project, it is indeed a very nice job.
I am currently trying to set up a configuration with a Asterisk PBX server and I have 2 questions:
1/ RTP configuration 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 ?
sip local 0.0.0.0 5069 remote 127.0.0.1 5060
2/ codec issue 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 ---> INVITE sip:899@127.0.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.184.10.162:5069;rport;branch=z9hG4bKpe1UXZU1amUja Max-Forwards: 70 From: sip:422@0.0.0.0:5069;tag=vyQKX32r72ZyQ To: sip:899@127.0.0.1:5060 Call-ID: cd65c5a0-fdbf-1238-51a9-000c29cfd753 CSeq: 949096397 INVITE Contact: sip:10.184.10.162:5069 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!
Thanks for your help Laurent
On 20/04/2020 11:02, Laurent Kza wrote:
Hi all,
Hi Laurent, it's really a very very long time since I have used Asterisk, but I can possibly help you to work through this.
1/ RTP configuration
Just to be clear, you are using the full stack, osmo-bsc/msc/mgw ? or the osmo-nitb?
What is your BTS?
I was wondering if there is any possibility to send the RTP flow to an
address which is not localhost ?
It is signaled in the SDP, if you are originating the call from asterisk, then it's an asterisk parameter somewhere. If it's a mobile to mobile call, the B-leg is still "originating" from Asterisk.
When you say "localhost instead of your server address", can you clarify, how many "servers" (be they VMs or whatever) are involved here? Just the one?
Maybe you mean your public (or private) address on a network card? To my mind, "localhost" is a "server address" certainly in the case of an osmo-mgw <--> asterisk stream on the same box, then it would be.
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!
This looks to me like Asterisk is pretty clearly saying it does not support the GSM codec, or does not have this enabled in configuration somewhere.
BTW, rather than looking at SIP log on the console, I highly recommend you use sngrep, it gives you a visual representation of SIP messages and media flows that makes every some much more clear!
K.
-
Keith -
Laurent Kza