Bug #3650
closedMSC is not sending a payload_type to MNCC?
0%
Description
The SDP created by osmo-sip-connector is invalid:
a=rtpmap:0 GSM/8000
Freeswitch rejects this.
Looks like the MSC is sending 0 for payload_type.
See mncc.c:check_rtp_create() in the sip cxtr
This works with legacy nitb:
You get this line in sip-connector's debug log:
mncc.c:393 RTP cnt leg(5010) ip(172.16.0.15), port(16412) pt(3) ptm(768)
With osmo-msc it will be:
mncc.c:393 RTP cnt leg(5010) ip(172.16.0.15), port(16412) pt(0) ptm(768)
note pt(0)
Related issues
Updated by neels about 5 years ago
let me copy the dirty hack mentioned on the ML here for later reference
(I haven't tested but wanted to find this if I need it.)
a quick and dirty hack for the osmo-sip-connector, to (probably) get your calls running through FreeSwitch: Hardcode override the pt in sdp_create_file() in sdp.c by adding other->payload_type = 98; (or for full rate GSM, it would be other->payload_type = 3;) somewhere in the top of that function, at line 170 for example, here: http://git.osmocom.org/osmo-sip-connector/tree/src/sdp.c#n170)
Updated by keith about 5 years ago
neels wrote:
For 35c3 congress, this would be interesting to clarify...
#3518 would also bite us at congress methinks..
and there's also a workaround:
https://gerrit.osmocom.org/#/c/osmo-sip-connector/+/11194/
Updated by keith about 5 years ago
Alternative:
In libmsc/gsm_04_08_cc.c (from line 1690), do:
/* FIXME: This has to be set to some meaningful value,
* before the MSC-Split, this value was pulled from
* lchan->abis_ip.rtp_payload */
uint32_t payload_type = 3;
Updated by fixeria almost 5 years ago
- Related to Bug #3724: Wrong media format used in SIP INVITE causes one-way audio added
Updated by keith about 4 years ago
- Related to Bug #1683: osmo-sip-connector: Implement codec selection / move codec selection to osmo-msc added
Updated by keith about 4 years ago
- Status changed from New to Closed
As this is being resolved in #1683, I'll close this issue.
