Bug #3650
closed
MSC is not sending a payload_type to MNCC?
Added by keith over 5 years ago.
Updated over 4 years ago.
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)
For 35c3 congress, this would be interesting to clarify...
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)
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;
- Related to Bug #3724: Wrong media format used in SIP INVITE causes one-way audio added
- Related to Bug #1683: osmo-sip-connector: Implement codec selection / move codec selection to osmo-msc added
- Status changed from New to Closed
As this is being resolved in #1683, I'll close this issue.
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