Hello,
I am planning to release Kamailio v4.3.7 very soon, likely tomorrow,
based on the last version of branch 4.3. This should mark the end of
official maintenance for branch 4.3, so if no major regression
discovered in following few weeks, it will be the last release in 4.3
series. If there is something you know it needs to be pushed in branch
4.3, write back to mailing list.
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com
<!--
Kamailio Project uses GitHub Issues only for bugs in the code or feature requests.
If you have questions about using Kamailio or related to its configuration file,
ask on sr-users mailing list:
* http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
If you have questions about developing extensions to Kamailio or its existing
C code, ask on sr-dev mailing list
* http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
Please try to fill this template as much as possible for any issue. It helps the
developers to troubleshoot the issue.
If you submit a feature request (or enhancement), you can delete the text of
the template and only add the description of what you would like to be added.
If there is no content to be filled in a section, the entire section can be removed.
You can delete the comments from the template sections when filling.
You can delete next line and everything above before submitting (it is a comment).
-->
### Description
<!--
Explain what you did, what you expected to happen, and what actually happened.
-->
When making end-to-end IMS voice call over IPv6, Kamailio PCSCF doesn't send the Sip Invite to callee. I saw the following error in PCSCF logs:
4(32107) DEBUG: <core> [parser/parse_uri.c:1258]: parse_uri(): parse_uri: bad char ':' in state 2 parsed: <sip:FC00:1234> (13) / <sip:FC00:1234:3:2:0:0:0:0:5060> (30)
4(32107) ERROR: tm [ut.h:254]: uri2dst2(): ERROR: uri2dst: bad_uri: [sip:FC00:1234:3:2:0:0:0:0:5060]
4(32107) ERROR: tm [t_fwd.c:1737]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches
4(32107) DEBUG: tm [t_funcs.c:331]: t_relay_to(): t_forward_nonack returned error -478 (-478)
4(32107) DEBUG: tm [t_funcs.c:348]: t_relay_to(): -478 error reply generation delayed
I saw a similar issue in your database: nat_traversal: Builds wrong URI for IPv6, bad URI parsing #362.
Have you fixed this issue?
### Troubleshooting
To me, sip:FC00:1234:3:2:0:0:0:0:5060 should be sip:[FC00:1234:3:2:0:0:0:0]:5060. After I added the workarround to change it to sip:[FC00:1234:3:2:0:0:0:0]:5060, PCSCF can send the Sip Invite to callee and the IMS voice call can be established.
#### Reproduction
<!--
If the issue can be reproduced, describe how it can be done.
-->
I can reproduce this issue using two soft phones: Linphone and Jitsi.
#### Debugging Data
<!--
If you got a core dump, use gdb to extract troubleshooting data - full backtrace,
local variables and the list of the code at the issue location.
gdb /path/to/kamailio /path/to/corefile
bt full
info locals
list
If you are familiar with gdb, feel free to attach more of what you consider to
be relevant.
-->
```
(paste your debugging data here)
```
#### Log Messages
<!--
Check the syslog file and if there are relevant log messages printed by Kamailio, add them next, or attach to issue, or provide a link to download them (e.g., to a pastebin site).
-->
```
(paste your log messages here)
```
4(32107) DEBUG: <core> [parser/parse_uri.c:1258]: parse_uri(): parse_uri: bad char ':' in state 2 parsed: <sip:FC00:1234> (13) / <sip:FC00:1234:3:2:0:0:0:0:5060> (30)
4(32107) ERROR: tm [ut.h:254]: uri2dst2(): ERROR: uri2dst: bad_uri: [sip:FC00:1234:3:2:0:0:0:0:5060]
4(32107) ERROR: tm [t_fwd.c:1737]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches
4(32107) DEBUG: tm [t_funcs.c:331]: t_relay_to(): t_forward_nonack returned error -478 (-478)
4(32107) DEBUG: tm [t_funcs.c:348]: t_relay_to(): -478 error reply generation delayed
#### SIP Traffic
<!--
If the issue is exposed by processing specific SIP messages, grab them with ngrep or save in a pcap file, then add them next, or attach to issue, or provide a link to download them (e.g., to a pastebin site).
-->
```
(paste your sip traffic here)
```
INVITE sip:bob@[fc00:1234:3:2:0:0:0:0]:5060;transport=udp;registering_acc=ims_mnc001_mcc001_3gppnetwork_org SIP/2.0
Record-Route: <sip:mt@[FC00:1234:1:0:0:0:0:47];lr=on;ftag=be2790bd;did=557.1972>
Route: <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org:5060;nat=yes;received=sip:FC00:1234:3:2:0:0:0:0:5060;lr>
Record-Route: <sip:mo@[FC00:1234:1:0:0:0:0:47];lr=on;ftag=be2790bd;did=557.1972>
Record-Route: <sip:mo@[FC00:1234:1:0:0:0:0:45];lr=on;ftag=be2790bd;nat=yes;did=557.f271>
Call-ID: 16a983b6edc2370ad08370112a800d98@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
From: "alice" <sip:alice@ims.mnc001.mcc001.3gppnetwork.org>;tag=be2790bd
To: <sip:bob@ims.mnc001.mcc001.3gppnetwork.org>
Via: SIP/2.0/UDP [FC00:1234:1:0:0:0:0:47];branch=z9hG4bKc87.6294094b4129fdf733069ffd6625d303.0
Via: SIP/2.0/UDP [FC00:1234:1:0:0:0:0:46];branch=z9hG4bKc87.66958ffdcce62f80024bf5913086e312.1
Via: SIP/2.0/UDP [FC00:1234:1:0:0:0:0:47];branch=z9hG4bKc87.5d9a7aae58830978432d73b25057fb39.0
Via: SIP/2.0/UDP [FC00:1234:1:0:0:0:0:45];branch=z9hG4bKc87.b56bcd8f177f65d2b1ec20ccca46519a.0
Via: SIP/2.0/UDP [fc00:1234:3:1:0:0:0:0]:5060;rport=5060;branch=z9hG4bK-363237-5db2dc2ac95e17a0c589ae29eb75a0ce
Max-Forwards: 66
Contact: "alice" <sip:alice@[fc00:1234:3:1:0:0:0:0]:5060;transport=udp;registering_acc=ims_mnc001_mcc001_3gppnetwork_org>
User-Agent: Jitsi2.10.5550Linux
Content-Type: application/sdp
Content-Length: 910
X-RTP: mo
P-Asserted-Identity: <sip:alice@ims.mnc001.mcc001.3gppnetwork.org>
v=0
o=alice-jitsi.org 0 0 IN IP6 fc00:1234:3:1:0:0:0:0
s=-
c=IN IP6 fc00:1234:3:1:0:0:0:0
t=0 0
m=audio 5056 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 101
a=rtpmap:96 opus/48000/2
a=fmtp:96 usedtx=1
a=ptime:20
a=rtpmap:97 SILK/24000
a=rtpmap:98 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:100 speex/32000
a=rtpmap:102 speex/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 speex/8000
a=rtpmap:101 telephone-event/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics
m=video 5058 RTP/AVP 105 99
a=recvonly
a=rtpmap:105 H264/90000
a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:105 send * recv [x=[1:1376],y=[1:883]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send * recv [x=[1:1376],y=[1:883]]
### Possible Solutions
<!--
If you found a solution or workaround for the issue, describe it. Ideally, provide a pull request with a fix.
-->
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
(paste your output here)
```
version: kamailio 5.0.0-dev5 (x86_64/linux)
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 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_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 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 on 21:48:42 Feb 21 2017 with gcc 5.4.0
* **Operating System**:
<!--
Details about the operating system, the type: Linux (e.g.,: Debian 8.4, Ubuntu 16.04, CentOS 7.1, ...), MacOS, xBSD, Solaris, ...;
Kernel details (output of `uname -a`)
-->
```
(paste your output here)
```
Distributor ID: Ubuntu
Description: Ubuntu 16.04.2 LTS
Release: 16.04
Codename: xenial
--
You are receiving this because you are subscribed to this thread.
Reply to this email directly or view it on GitHub:
https://github.com/kamailio/kamailio/issues/1136