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Bug #3650

MSC is not sending a payload_type to MNCC?

Added by keith 2 months ago. Updated 29 days ago.

Status:
New
Priority:
Normal
Assignee:
Category:
-
Target version:
-
Start date:
10/12/2018
Due date:
% Done:

0%

Resolution:

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

Related to osmo-sip-connector - Bug #3724: Wrong media format used in SIP INVITE causes one-way audioNew2018-12-11

History

#1 Updated by laforge 2 months ago

  • Assignee set to dexter

#2 Updated by neels about 1 month ago

For 35c3 congress, this would be interesting to clarify...

#3 Updated by neels about 1 month 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)

#4 Updated by keith about 1 month 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/

#5 Updated by keith 29 days 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;

#6 Updated by fixeria 4 days ago

  • Related to Bug #3724: Wrong media format used in SIP INVITE causes one-way audio added

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