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FS#100 - Assignment operators don't work
User who did this - Alex Hermann (axlh)
http://sip-router.org/tracker/index.php?do=details&task_id=100
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Changes to example files for PCSCF in misc/examples/ims to make funcitional with current stable version.
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#### Description
Changes to example files for PCSCF in misc/examples/ims to make funcitional with current stable version.
Changes:
- Loading IPsec module prior to IMS Usrloc PCSCF (Now required)
- removed modparam("ims_usrloc_pcscf", "hashing_type", 2) from example (This parameter was removed some time ago)
- Fix to formatting of single MySQL connection to work in current version
- Bind to any IP by default
- Dispatcher parameters only loaded if required
You can view, comment on, or merge this pull request online at:
https://github.com/kamailio/kamailio/pull/2203
-- Commit Summary --
* misc: examples: IMS PCSCF kamailio.cfg update
* misc: examples: IMS PCSCF pcscf.cfg update
-- File Changes --
M misc/examples/ims/pcscf/kamailio.cfg (6)
M misc/examples/ims/pcscf/pcscf.cfg.sample (12)
-- Patch Links --
https://github.com/kamailio/kamailio/pull/2203.patchhttps://github.com/kamailio/kamailio/pull/2203.diff
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Dear Gang
Possibly @oej could provide more in-depth information as he has witnessed this issue.
Usually the user of the from URI is the phone number displayed at the destination. There are situations where this phone number is translated.
As example. In Switzerland, the user is used to see numbers in a local format. National number starting with 0 and international numbers with 00 but on interconnection between telcos, e164 is used.
So basically when a call is sent to a customer '+41' is replaced by '0' and '+' is replaced by '00'.
Let's start with an example From: header:
`From: "Maurice Moss" <sip:+41991234567@example.com>;user=phone`
So shortly before the call is sent out to the location of the registered CPE, this is done:
```
if ($fU =~ "^\+41") {
$fU = "0" + $(fU{s.substr,3,0});
} else if ($fU = ~ "^\+") {
$fU = "00" + $(fU{s.substr,1,0});
}
```
What is sent to the CPE now looks like this:
`From: "Maurice Moss" <sip:0991234567@example.com>;user=phone`
Now we hit an error like 486 BUSY and the destination has call forwarding active to a mobile phone on another TSP. So we have to send the call out back the IC and numbers need to be translated back to e164.
We handle this in a failure route, which in turn could trigger a branch route.
So we revert the number back to e164:
`$fU = "+41" + $(fU{s.substr,1,0});`
Expected outcome:
`From: "Maurice Moss" <sip:+41991234567@example.com>;user=phone`
Observed outcome:
`From: "Maurice Moss" <sip:0991234567+41991234567@example.com>;user=phone`
So setting $fU more than once is appending to the user element of the From header URI.
This behavior has not been found in any documentation.
I have been working around most of the issues by making sure I change $fU (and $tU) at the latest possible time and only once. But in the case described above, I have not been able to come up with a work-around yet.
I also can't think of any benefit of the way those PV are handled or any harm that could be done, to handle them differently and make the last 'write' overwrite and previous value, instead of appending.
Thank you for looking into this.
-Benoît-
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### Description
In some situations the other side UA must reply with an inactive media stream. E.g. if video was added but not supported by the other side. The UAS then must answer with an m=video 0 RTP/.. line.
It seams that the video section must contain a c= line with an IP address for rtpengine to function. If there is no c line (no IP address) the SDP body cannot be parsed and thus no RTP proxy is invoked.
### Troubleshooting
Not easy to reproduce since the answer must bei "wrong". Is it correct to reply an SDP body (media = video) like this?
```
m=video 0 RTP/AVPF 96
a=label:1
a=inactive
a=mid:1
```
Maybe not (but Audiocodes SBC lates LTS version does it) - so we are dependent on other side now that this service works.
#### Reproduction
reInvite with Video to a UA that has no video Support (e.g. Bria -> Snom). Drop the c line in the video part with some textops in 200 OK (just to reproduce of course).
A -> B, connect the call
A -> + video, B has no video support
A presses hold
#### Log Messages
Incoming 200 OK after doing hold with an inactive video:
```
v=0
o=root 436040161 309284577 IN IP4 1.2.3.5
s=call
t=0 0
m=audio 14200 RTP/AVPF 9 8 126
c=IN IP4 1.2.3.5
a=ptime:20
a=recvonly
a=mid:0
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-15
m=video 0 RTP/AVPF 96
a=label:1
a=inactive
a=mid:1
Mar 27 17:52:22 c5p05es2-1 /usr/sbin/kamailio[2938205]: ERROR: <core> [core/parser/sdp/sdp.c:490]: parse_sdp_session(): can't find media IP in the message
Mar 27 17:52:22 c5p05es2-1 /usr/sbin/kamailio[2938205]: INFO: <script>: >>> Sending Reply: 200 OK (1.2.3.4:443 -> 100.108.48.38:65251)
```
Compare to working SDP parsing:
```
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-15
m=video 0 RTP/AVP 96
c=IN IP4 1.2.3.5
a=inactive
a=mid:1
Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: ONREPLY_ROUTE[FORWARD]
Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: ROUTE[RTPP_REPLY]
Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > 200 with video
Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > 200 with inactive video
Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > Remove inactive video
Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: > Answer to WebRTC client: 200 - qijpuibkv6ufhnd282ks
Mar 27 18:00:17 c5p05es2-1 /usr/sbin/kamailio[2942881]: INFO: <script>: >>> Sending Reply: 200 OK (91.237.65.14:443 -> 80.108.48.38:65388)
```
### Possible Solutions
Tried to fix in script:
```
if(sdp_with_media("video")) xlog("L_INFO", "> $rs with video");
if(sdp_with_active_media("video")) xlog("L_INFO", "> $rs with active video");
if(!sdp_with_active_media("video")) xlog("L_INFO", "> $rs with inactive video");
if(sdp_with_media("video") && !sdp_with_active_media("video")) {
# try fix for RMT#60189
xlog("L_INFO", "> Remove inactive video");
sdp_remove_media("video");
msg_apply_changes(); # needed?
}
```
But since sdpops cannot parse it cannot remove the buggy media, too. See the log: non of the xlog is logging.
If there is inactive media the c line is not mandatory. The parser should accept this by adding a default IP of 0.0.0.0, if needed
Doing tricks with textops is not a easy nor performant solution to accept this wrong SDP from other parties.
### Additional Information
```
version: kamailio 5.6.4 (x86_64/linux)
flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED
ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled with gcc 10.2.1
```
* **Operating System**:
Debian Bullseye
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From the build log
```
CC (gcc) [M kazoo.so] kazoo.o
CC (gcc) [M kazoo.so] kz_amqp.o
kz_amqp.c: In function 'kz_amqp_connection_open_ssl':
kz_amqp.c:857:17: warning: 'amqp_set_initialize_ssl_library' is deprecated [-Wdeprecated-declarations]
857 | amqp_set_initialize_ssl_library(1);
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
In file included from kz_amqp.c:37:
/usr/include/rabbitmq-c/ssl_socket.h:233:16: note: declared here
233 | void AMQP_CALL amqp_set_initialize_ssl_library(amqp_boolean_t do_initialize);
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
CC (gcc) [M kazoo.so] kz_fixup.o
CC (gcc) [M kazoo.so] kz_hash.o
```
Functions description in the header file
https://github.com/alanxz/rabbitmq-c/blob/v0.13.0/include/rabbitmq-c/ssl_so…
```
/**
* Sets whether rabbitmq-c will initialize OpenSSL.
*
* \deprecated Since v0.13.0 this is a no-op. OpenSSL automatically manages
* library initialization and uninitialization.
*
* OpenSSL requires a one-time initialization across a whole program, this sets
* whether or not rabbitmq-c will initialize the SSL library when the first call
* to amqp_ssl_socket_new() is made. You should call this function with
* do_init = 0 if the underlying SSL library is initialized somewhere else
* the program.
*
* Failing to initialize or double initialization of the SSL library will
* result in undefined behavior
*
* By default rabbitmq-c will initialize the underlying SSL library.
*
* NOTE: calling this function after the first socket has been opened with
* amqp_open_socket() will not have any effect.
*
* \param [in] do_initialize If 0 rabbitmq-c will not initialize the SSL
* library, otherwise rabbitmq-c will initialize the
* SSL library
*
* \since v0.4.0
*/
```
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