[sr-dev] [kamailio/kamailio] IPv6 parse_uri error (#1136)
davidxbwang
notifications at github.com
Tue May 23 21:30:47 CEST 2017
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### Description
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Explain what you did, what you expected to happen, and what actually happened.
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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
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I can reproduce this issue using two soft phones: Linphone and Jitsi.
#### Debugging Data
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```
(paste your debugging data here)
```
#### Log Messages
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```
(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 at 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 at 0:0:0:0:0:0:0:0
CSeq: 1 INVITE
From: "alice" <sip:alice at ims.mnc001.mcc001.3gppnetwork.org>;tag=be2790bd
To: <sip:bob at 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 at 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
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### 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**:
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```
(paste your output here)
```
Distributor ID: Ubuntu
Description: Ubuntu 16.04.2 LTS
Release: 16.04
Codename: xenial
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