There are allocated bits in conn->end.codecs[], free them.
This is not fixing a memleak, since mgcp_rtp_conn_cleanup() is currently only
called from mgcp_conn_free(), which soon after frees the conn; the conn serves
as talloc parent for the codec strings freed in this patch.
The rationale: it is better style to explicitly free them, to also guard
against future callers of mgcp_rtp_conn_cleanup() which might expect complete
cleanup.
Change-Id: Ic471107ce6e94d9ce582d887429c744ff93e3053
In codec_set(), for each 'goto error', log the specific error cause.
Also add a TODO and a FIXME comment about inventing dynamic payload type
numbers.
Change-Id: I0b44b574c814882b6f8ae7cd738a6f481cd721fd
In codecs_same(), do not compare the complete audio_name. The parts of it are
already checked individually:
- subtype_name ("AMR"),
- rate ("8000"; defaults to 8000 if omitted) and
- channels ("1"; defaults to 1 if omitted)
So by also checking the complete audio_name, we brushed over the match of
implicit "/8000" and "/8000/1", which otherwise works out fine.
As a result, translating payload type numbers in RTP headers now also works if
one conn of an endpoint set an rtpmap with "AMR/8000" and the other conn set
"AMR/8000/1".
It seems to me that most PBX out there generate ptmaps omitting the "/1", so
fixing this should make us more interoperable with third party SDP.
See IETF RFC4566 section 6. SDP Attributes:
For audio streams, <encoding parameters> indicates the number
of audio channels. This parameter is OPTIONAL and may be
omitted if the number of channels is one, provided that no
additional parameters are needed.
Also allowing to omit the "/8000" is a mere side effect of this patch.
Omitting the rate does not seem to be specified in an RFC, but is logical for
audio codecs defined to require exactly 8000 set as rate (most GSM codecs).
Add tests in mgcp_test.c.
Change-Id: Iab00bf9a55b1847f85999077114b37e70fb677c2
The audio_name and subtype_name are allocated from talloc, so they need to be
freed before resetting the codec array. Use mgcp_codec_free() to ensure this.
Change-Id: I07f207dcb7ce66bbf3445a30af41e696677b384f
Both are used only in the same .c file, so make them static.
Move codec_set() guts into codec_add(): codec_set is only called by codec_add.
If codec_set were left separate, it'd look like the codec_init() is a bug and
lacks a codec_free() first. When looking at the entire context in codec_add(),
it becomes obvious that codec_init() should be called, not codec_free(),
because it is populating a previously unused entry.
Preparation to fix a memleak in a conn's codec list.
Change-Id: I120cab0a352a1e7b31c8f9c720c47b2c291311d7
If mgcp_send() runs a transcoder loop, break the loop if rfc5993_hr_convert()
or amr_oa_bwe_convert() return with error. Possibly fixes an infinite loop
situation for erratic packets? (Didn't check for that in detail.)
Change-Id: Iba115a0b1d74e7cefba5dcdd777e98ddea9eba8c
Remove various OSMO_ASSERT() on size of incoming packets. Doing an assert on
incoming data is a DoS attack vector, absolute no-go. Instead, return -EINVAL
and keep running.
Change some return values to be able to distinguish successful operation from
invalid RTP sizes. In rtp_data_net(), make sure to return negative if the RTP
packet was invalid.
Some of the error return codes implemented here will only be used in upcoming
patch Iba115a0b1d74e7cefba5dcdd777e98ddea9eba8c.
Change-Id: I6bc6ee950ce07bcc2c585c30fad02b81153bdde2
The name 'cmp' implies a return value of -1, 0, 1 to indicate smaller, match or
larger. Since this function returns bool, it should not be named with 'cmp'.
Change-Id: I2d41b1a32300e295551e85d3f9ab82dd2b0e86b8
Return variable specified by strtoul() is "unsigned long int". If
"unsigned int" is used, according to Coverity the return value can never
be ULONG_MAX:
CID 202173: Integer handling issues (CONSTANT_EXPRESSION_RESULT)
"pt == 18446744073709551615UL /* 9223372036854775807L * 2UL + 1UL */" is always false regardless of the values of its operands. This occurs as the logical second operand of "&&".
Furthermore, PT is 7 bit in RTP header [1], so let's avoid accepting
incorrect values.
[1] https://tools.ietf.org/html/rfc3550#section-5
Fixes: c5c1430a1c ("Catch unsigned integer MGCP parsing errors with strtoul")
Fixes: Coverity CID#202172
FIxes: Coverity CID#202173
Change-Id: Ice9eee6a252fab73dbab5ebf3cfc83c1b354fd08
MGCP RFC3435 (https://tools.ietf.org/html/rfc3435) states almost all
text has to be handled in a case-insensitive way, except SDP parts.
Related: OS#4001
Change-Id: I637cb20f0af4de33ebf6589b1aff260d57d03e7b
MGCP RFC3435 (https://tools.ietf.org/html/rfc3435) states almost all
text has to be handled in a case-insensitive way, except SDP parts.
Related: OS#4001
Change-Id: Ifc1b3bfe6ff6922df478cea89bbbb291b5fa5706
MGCP RFC3435 (https://tools.ietf.org/html/rfc3435) states almost all
text has to be handled in a case-insensitive way, except SDP parts.
Related: OS#4001
Change-Id: I7d1e55faddafa3c3093d38513d4a434ecf5ea5bd
Otherwise it would not catch a duplicate if first the param is
introduced in upper case and later in lower case, or the other way
around.
MGCP RFC3435 (https://tools.ietf.org/html/rfc3435) states almost all
text has to be handled in a case-insensitive way, except SDP parts.
Related: OS#4001
Change-Id: I254bfa3a2d2562441ca3a576cc8e1e7967d9c495
MGCP RFC3435 (https://tools.ietf.org/html/rfc3435) states almost all
text has to be handled in a case-insensitive way, except SDP parts.
Related: OS#4001
Change-Id: I51dc1cdcbe2a5587769335fbecb5039ef22cae5d
MGCP RFC3435 (https://tools.ietf.org/html/rfc3435) states almost all
text has to be handled in a case-insensitive way, except SDP parts.
Related: OS#4001
Change-Id: I4da93dfc69b5585a197a7e201a1afb72c2f97030
MGCP RFC3435 (https://tools.ietf.org/html/rfc3435) states almost all
text has to be handled in a case-insensitive way, except SDP parts.
Related: OS#4001
Change-Id: I48252415f9d0cd985ad097f334aa4c1665f52511
MGCP RFC3435 (https://tools.ietf.org/html/rfc3435) states almost all
text has to be handled in a case-insensitive way, except SDP parts.
Related: OS#4001
Change-Id: Ic28a5eacc4c441d68e8a20d2743956ab2e01125d
It can be used together with LCLS, just make sure to also enable
keep-alive packets.
In OS#3429 it was pointed out, that during LCLS the media path remains
active but is not used. Without any traffic flowing, this looks like a
timed out connection and so it will be killed if conn-timeout is set.
However, OsmoBSC and OsmoMSC have an option to enable RTP keep-alive
packets (through libosmo-mgcp, originally intended to keep connections
behind NAT open). If that option is enabled, the keep-alive packets
should also prevent the conn-timeout.
Related: OS#3783
Change-Id: Ib4d19104d558a26a444a80fb36f4b7b33bc5cc59
VTY command to disable conn-timeout again, after it has been enabled.
conn-timeout was introduced in [1].
[1] Change-Id I18886052e090466f73829133c24f011806cc1fe0.
Change-Id: I7dee7dafaaf4bb93fd692ea06b52b9e012beac6d
Move code in RTP specific path to generic dispatch_rtp_cb. This way
loopback logic is applied both for Osmux and RTP connections.
Change-Id: Ia30f5a14f150e4d151eac4d1046ea834f1685a5f
That MGCP_DUMMY_LOAD is an old hack prior to Osmux spec update, and it's
not nice since it cannot be 100% distinguished from a usual AMR ft
frame.
Let's use the correct DUMMY ft type and build it according spec. Allow
handling differently the old format for a while until we are sure no old
implementations (like bsc-nat) exist sending that kind of message.
Change-Id: Ib17d20b87b28aade49ba60519b56a96e694819af
Remove old BTS/NET no longer in use and meaningless. Use new osmo-mgw
APIs to inject payload RTP<->Osmux on the correct socket and conn.
Change-Id: I60b6ba3ffdc74efff945ba13a0b736798bdf5d8c
Previously the local one was used but nobody cared because probably
everybody was using default 1984 on different IP addresses.
Change-Id: I01e590465fa247185d74103578681e9041249099
During MDCX state is already changed to ACTIVATING but we still want to
send the local CID back to announce that we still use same local CID.
Change-Id: If182a48743ebe03f97caf9034e49b9947014bdf9
We also update code to allow setting up RTP related fields to succeed
during CRCX. We also update code to allow setting up RTP related fields to
succeed during CRCX.
Change-Id: Ia6e723d9a28ba38fc3382a4fb35ea6e5bab30c09
osmux_xfrm_input_open_circuit returns 0 on success and -1 on error.
Confusion comes from that function being implemented by calling
osmux_batch_add_circuit which returns NULL on error.
cherry-picked from: openbsc.git ac1b03c8e59408336d07527e2597171cb99ed654.
Change-Id: Iba018aa57901642ea4c486526a973fe6023e10cf
* Cleanup naming to make it far more clear
* Drop 2 variables holding CID values (allocated_cid, cid), in favour of
1 value holdinf the value and one bool stating whether the value is
used.
* Change conn_osmux_allocate_cid to allow allocating either next
available CID or a specific one, in preparation for forthcoming patches.
This commit can be merged straight away because anyway osmux cannot be
enabled in current status (blocked by vty config) and
(conn_)osmux_allocate_cid was/is not called anywhere. However, it helps
improving code base for future re-introduction of Osmux as it is
envisioned.
Change-Id: I737a248ac6c74add8e917fe2e2f36779d0f1d685
Pau Espin
Neels Hofmeyr
Harald Welte
Oliver Smith