Support #4137
closed"can not patch PT because no suitable egress codec was found" = One way audio
0%
Description
I'm finally in the process of converting my working NITB instance over to the newer osmo-splits.
I am running the Stable Binary Packages.
The trouble is that i'm getting one way audio when sending calls to a local Asterisk server (Originating GSM caller hears silence. Terminated SIP caller can hear GSM user).
osmo-mgw spams these messages:
% <0000> mgcp_network.c:726 endpoint:0x0 can not patch PT because no suitable egress codec was found.
% <0000> mgcp_network.c:726 endpoint:0x0 can not patch PT because no suitable egress codec was found.
% <0000> mgcp_network.c:726 endpoint:0x0 can not patch PT because no suitable egress codec was found.
% <0000> mgcp_network.c:726 endpoint:0x0 can not patch PT because no suitable egress codec was found.
% <0000> mgcp_network.c:726 endpoint:0x0 can not patch PT because no suitable egress codec was found.
% <0000> mgcp_network.c:726 endpoint:0x0 can not patch PT because no suitable egress codec was found.
In reviewing the SIP Debug within Asterisk, I can see GSM in the SDP. I also see the same thing in osmo-sip-connector when setting the Sofia SIP debug very high.
As a test, I set MSC to use the Internal MNCC and i'm able to get two way audio between GSM Handsets -- however, as the end game is to use my Asterisk PBX, this is obviously not the best solution.
I've done my best to search, came across this: https://osmocom.org/issues/3550 which seems similar, but had no resolution explanation. That issue states it was an Asterisk NAT issue. For testing I toggled the NAT settings, no change.
Attaching all configs.
Am I missing something?
Files
Updated by laforge over 4 years ago
- Assignee set to dexter
- Priority changed from Normal to Low
Updated by the_toph over 4 years ago
Is there any additional information I could provide to assist in troubleshooting this problem?
Updated by dexter over 4 years ago
- Status changed from New to In Progress
I had a look at your BSC config and I see that you allow the following codecs:
codec-list fr1 hr1 fr2 fr3 hr3
I would suggest to limit your configuration to fr1 only. The reason for this is that there are still negotiation problems with the MNCC interface, which basically limits everything to FR1. I have the osmo-splits running with Asterisk, so I can confirm that it works in general.
It would also be good to have a trace from that includes BSSMAP, MGCP, SIP and possibly RTP. Possibly there is some problem negotiating the codec with the MGW.