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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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Message ID: <kamailio/kamailio/issues/3165(a)github.com>
### 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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When relaying this `200 OK` repeatedly with calls to `rtpengine_answer()` failing because all RTPEngines in a set are unreachable...
```
Dec 8 06:24:38 gw kamailio[4823]: ERROR: rtpengine [rtpengine.c:2960]: select_rtpp_node(): rtpengine failed to select new for calllen=36 callid=8279a4ea-151e-48de-a23c-5840104126de
```
```
SIP/2.0 200 OK
v:SIP/2.0/UDP 10.150.20.20;branch=z9hG4bK03e.e4da444e627bb575ef4f46151729a1fa.0,SIP/2.0/UDP 10.151.20.108;received=10.151.20.108;rport=5060;branch=z9hG4bKKQyrBKpp04X0K
Record-Route:<sip:10.150.20.20;lr;ftag=emHm2apy95jFe;rtp_relay=1;rtp_group=0>
f:"14045551212"<sip:14045551212@10.151.20.108>;tag=emHm2apy95jFe
t:<sip:16782001111@internal.evaristesys.com>;tag=tHQt0rDcF7UHK
i:8279a4ea-151e-48de-a23c-5840104126de
CSeq:60686471 INVITE
m:<sip:16782001111@10.160.10.55:5060;transport=udp>
User-Agent:VoiceAppServer
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REFER,NOTIFY
k:path,replaces
u:talk,hold,conference,refer
c:application/sdp
Content-Disposition:session
l:257
v=0
o=DM 1670428117 1670428118 IN IP4 10.160.10.55
s=DM
c=IN IP4 10.160.10.55
t=0 0
m=audio 52418 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp:52419 IN IP4 10.160.10.55
```
Repeat INVITEs eventually result in a crash. This backtrace is taken from a different server but identical scenario:
```
#0 t_should_relay_response (Trans=0x7f4aee56ba80, new_code=200, branch=0, should_store=0x7ffd8b8bbba0, should_relay=0x7ffd8b8bbba4, cancel_data=0x7ffd8b8bbe40,
reply=0x7f4afe40df70) at t_reply.c:1285
1285 && !(inv_through && Trans->uac[branch].last_received<300)) {
Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.192.el6.x86_64 jansson-2.11-1.el6.x86_64 keyutils-libs-1.4-5.el6.x86_64 krb5-libs-1.10.3-57.el6.x86_64 libcom_err-1.41.12-22.el6.x86_64 libev-4.03-3.el6.x86_64 libselinux-2.0.94-7.el6.x86_64 libunistring-0.9.3-5.el6.x86_64 openssl-1.0.1e-48.el6_8.3.x86_64 sqlite-3.6.20-1.el6_7.2.x86_64 zlib-1.2.3-29.el6.x86_64
(gdb) where
#0 t_should_relay_response (Trans=0x7f4aee56ba80, new_code=200, branch=0, should_store=0x7ffd8b8bbba0, should_relay=0x7ffd8b8bbba4, cancel_data=0x7ffd8b8bbe40,
reply=0x7f4afe40df70) at t_reply.c:1285
#1 0x00007f4afd73ab7d in relay_reply (t=0x7f4aee56ba80, p_msg=0x7f4afe40df70, branch=0, msg_status=200, cancel_data=0x7ffd8b8bbe40, do_put_on_wait=1)
at t_reply.c:1821
#2 0x00007f4afd740c38 in reply_received (p_msg=0x7f4afe40df70) at t_reply.c:2558
#3 0x00000000004b5d6a in do_forward_reply (msg=0x7f4afe40df70, mode=0) at core/forward.c:747
#4 0x00000000004b78c9 in forward_reply (msg=0x7f4afe40df70) at core/forward.c:852
#5 0x000000000054982b in receive_msg (
buf=0xb27700 "SIP/2.0 200 OK\r\nv:SIP/2.0/UDP xxx;branch=z9hG4bK9915.dcfe6d03cb21e6c94a6705adb095c566.0,SIP/2.0/UDP xxx;received=xxx;rport=5060;branch=z9hG4bKZmZp0NZv3FvQB\r\nRecord-Rout"..., len=984, rcv_info=0x7ffd8b8bc6c0) at core/receive.c:434
#6 0x00000000006658ca in udp_rcv_loop () at core/udp_server.c:541
#7 0x000000000042500f in main_loop () at main.c:1655
#8 0x000000000042c5bb in main (argc=11, argv=0x7ffd8b8bcc88) at main.c:2696
```
Looks like in the crash frame, `Trans->uac` is `NULL`:
```
(gdb) frame 0
#0 t_should_relay_response (Trans=0x7f4aee56ba80, new_code=200, branch=0, should_store=0x7ffd8b8bbba0, should_relay=0x7ffd8b8bbba4, cancel_data=0x7ffd8b8bbe40,
reply=0x7f4afe40df70) at t_reply.c:1285
1285 && !(inv_through && Trans->uac[branch].last_received<300)) {
(gdb) print Trans->uac
$1 = (struct ua_client *) 0x0
```
I have seen this crash from time to time, though it is infrequent. It always seems to occur when there are problems reaching RTPEngine in a timely fashion.
This is on Kamailio 5.2.4:de9a03, which I know has long fallen off the back of the maintenance train. However, line 1386 of `t_reply.c` in `master:HEAD` shows that the code is no different now:
https://github.com/kamailio/kamailio/blob/master/src/modules/tm/t_reply.c#L…
Of course, it's possible that something upstream has changed so as to prevent this state. Otherwise, I suppose a null pointer safety check for `Trans->uac` is recommended. I'd be happy to submit a patch, just wasn't sure if the feeling was more that this should be resolved higher up the call stack.
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