6 May
2016
6 May
'16
5:08 a.m.
Hello,
I use kamailio set up an IMS on a cloud server in a public network, and have
alice and bob registered on the server using ImsDroid in a private network.
When I use alice to call bob, I got "User offline" error.
Below are .pcap file captured from 4060 port and
pcscf.log/icscf.log/scscf.log and all config files(in kamailio.zip)
4060.pcap <http://sip-router.1086192.n5.nabble.com/file/n148166/4060.pcap>
pcscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/pcscf.log>
icscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/icscf.log>
scscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/scscf.log>
kamailio.zip
<http://sip-router.1086192.n5.nabble.com/file/n148166/kamailio.zip>
My kamailio version and debian operating system version is like below:
root@iZ94lg22es7Z:~# kamailio -version
version: kamailio 4.4.0 (x86_64/linux) d4f23c
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: d4f23c
compiled on 13:02:46 May 5 2016 with gcc 4.9.2
root@iZ94lg22es7Z:~#
root@iZ94lg22es7Z:~# cat /etc/issue
Debian GNU/Linux 8 \n \l
root@iZ94lg22es7Z:~#
Can anybody give me some help?
Any suggestion will be appreciated!
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6 May
6 May
5:25 a.m.
Is this a NAT issue?
The network is like this：
alice--192.168.199.214 company ip--116.228.89.198 kamailio IMS(cloud
server)--120.76.118.128
bob--192.168.199.124
I can see INVITE was sent from 116.228.89.198(alice) to
120.76.118.128(kamailio IMS), then 120.76.118.128(kamailio IMS) reply a 100
trying to 116.228.89.198(alice), after that 120.76.118.128(kamailio IMS)
sent INVITE to 192.168.199.124(bob)
However, I did not captured bob's 183/180 or 200 OK.
What should I do to let it run properly?
<http://sip-router.1086192.n5.nabble.com/file/n148168/4060.jpg>
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5:31 a.m.
Is this a NAT issue?
The network is like this：
alice--192.168.199.214 company ip--116.228.89.198 kamailio IMS(cloud
server)--120.76.118.128
bob--192.168.199.124
I can see INVITE was sent from 116.228.89.198(company ip) to
120.76.118.128(kamailio IMS), then 120.76.118.128(kamailio IMS) reply a 100
trying to 116.228.89.198(company ip), after that 120.76.118.128(kamailio
IMS) sent INVITE to 192.168.199.124(private ip of bob), because
192.168.199.124 is a local ip address, I guess kamailio IMS in the public ip
address can't find a route to this address.
So, bob did not receive this INVITE message from alice.
Is my analysis right? What should I do to let the IMS run properly?
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5:32 a.m.
Is this a NAT issue?
The network is like this：
alice--192.168.199.214 company ip--116.228.89.198 kamailio IMS(cloud
server)--120.76.118.128
bob--192.168.199.124
I can see INVITE was sent from 116.228.89.198(company ip) to
120.76.118.128(kamailio IMS), then 120.76.118.128(kamailio IMS) reply a 100
trying to 116.228.89.198(company ip), after that 120.76.118.128(kamailio
IMS) sent INVITE to 192.168.199.124(private ip of bob), because
192.168.199.124 is a local ip address, I guess kamailio IMS in the public ip
address can't find a route to this address.
So, bob did not receive this INVITE message from alice.
Is my analysis right? What should I do to let the IMS run properly?
<http://sip-router.1086192.n5.nabble.com/file/n148172/4060.jpg>
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8:45 a.m.
hello;
try to open nat_travesal module on registrar server. your analyz is right in
my view. INVITE packets never reach to BOB.
cheers.
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8:50 a.m.
Thanks for your reply!
I found nat_traversal.so in /usr/local/lib64/kamailio/modules/ directory.
But didn't find a switch in pcscf.cfg or kamailio.cfg.
So how can I open nat_travesal module?
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8:59 a.m.
Hello,ycaner!
Thanks for your advice!
I searched nat_traversal in kamailio.cfg in PCSCF module, and found that I
should define WITH_NATPING.
After doing this, a new problem came. What does this line mean? What should
I configure to replace
"sip:keepalive@HOSTNAME" ? Is this HOSTNAME a turn/stun server?
modparam("nat_traversal", "keepalive_from",
*"sip:keepalive@HOSTNAME"*)
#!ifdef WITH_NATPING
# ----- nat_traversal params -----
# If another keepalive is wanted, this is the place
modparam("nat_traversal", "keepalive_interval", 20)
# If another method than NOTIFY is wanted:
modparam("nat_traversal", "keepalive_method", "OPTIONS")
# From?
modparam("nat_traversal", "keepalive_from",
"sip:keepalive@HOSTNAME")
# Where we store information about keep-alives:
modparam("nat_traversal", "keepalive_state_file",
"/var/run/kamailio/keepalive_state")
#!endif
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9:05 a.m.
Hello;
at registration route , you can use nat traversal function , there
is a sample on kamailio.cfg
that parameter is just adding from and request header for
information on OPTIONS packet. you can add your 120.XXX.XXX.XXX
Cheers.
2016-05-06 08:59 tarihinde 王欣鑫 [via SIP
Router] yazdı:
Hello,ycaner!
Thanks for your advice!
I searched nat_traversal in kamailio.cfg in PCSCF module, and
found that I should define WITH_NATPING.
After doing this, a new problem came. What does this line mean?
What should I configure to replace
"sip:keepalive@HOSTNAME" ? Is this HOSTNAME a turn/stun server?
modparam("nat_traversal", "keepalive_from",
"sip:keepalive@HOSTNAME")
#!ifdef WITH_NATPING
# ----- nat_traversal params -----
# If another keepalive is wanted, this is the place
modparam("nat_traversal", "keepalive_interval", 20)
# If another method than NOTIFY is wanted:
modparam("nat_traversal", "keepalive_method",
"OPTIONS")
# From?
modparam("nat_traversal", "keepalive_from",
"sip:keepalive@HOSTNAME")
# Where we store information about keep-alives:
modparam("nat_traversal", "keepalive_state_file",
"/var/run/kamailio/keepalive_state")
#!endif
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9:20 a.m.
Hello ycaner!
I did as you told me and finally it worked!
Here are my finall config files.
hss_config.zip
<http://sip-router.1086192.n5.nabble.com/file/n148179/hss_config.zip>
kamailio.zip
<http://sip-router.1086192.n5.nabble.com/file/n148179/kamailio.zip>
Thank you very much!!!
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9:56 a.m.
Hello,ycaner!
Now I use a real IMS user data to register this user(for example
18616278715) to China Unicom's IMS through kamailio.
I can register successsfully, and can make call using this number, the
callee can pick up phone. However
neither party can hear counter part.
So, this should be rtp problem. And I captured a .pcap file on port 5060
like below. In this, I see in SDP information
it says 192.168.199.124, it's a private address. So maybe the callee can't
setup a circuit to this ip, and thus no media was transfered?
Should I uncomment RTPPROXY_ADDRESS, as in my pcscf.cfg there are two lines:
# IP-Adress(es) of the RTP-Proxy
##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222 udp:10.1.27.217:22222"
##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222"
This is the .pcap file captured on port 5060.
5060.pcap <http://sip-router.1086192.n5.nabble.com/file/n148182/5060.pcap>
<http://sip-router.1086192.n5.nabble.com/file/n148182/5060.jpg>
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9:59 a.m.
Hello,ycaner!
Now I use a real IMS user data to register this user(for example
18616278715) to China Unicom's IMS through kamailio.
I can register successsfully, and can make call using this number, the
callee can pick up phone. However
neither party can hear counter part.
So, this should be rtp problem. And I captured a .pcap file on port 5060
like below. In this, I see in SDP information
it says 192.168.199.124, it's a private address. So maybe the callee can't
setup a circuit to this ip, and thus no media was transfered?
Should I uncomment RTPPROXY_ADDRESS, as in my pcscf.cfg there are two lines.
If I should do this, what ip and port should I use to replace
"udp:10.1.2.186:22222 udp:10.1.27.217:22222" ?
# IP-Adress(es) of the RTP-Proxy
##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222 udp:10.1.27.217:22222"
##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222"
This is the .pcap file captured on port 5060.
5060.pcap <http://sip-router.1086192.n5.nabble.com/file/n148183/5060.pcap>
<http://sip-router.1086192.n5.nabble.com/file/n148183/5060.jpg>
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10:08 a.m.
Hello;
you can use rtpproxy or rtpengine. you need to search how to use
on google. rtpproxy/rtpengine fixes sdp packets that has tons of
flags and there is a lot of option on sdpops module to change it.
for example; you can check with nat_uac_client(8) and fix sdp with
rptproxy_offer/rtpproxy_manage("cor") function.
there is 2-3 sample about it , here is kamailio doc how to use
functions.
http://www.kamailio.org/docs/modules/4.3.x/modules/rtpproxy.html#idp2564344
http://www.kamailio.org/docs/modules/4.3.x/modules/nathelper.html#nathelper…
-- samples ---
http://saevolgo.blogspot.com.tr/2013/08/rtpproxy-revisited-kamailio-40.html
http://www.fredposner.com/1457/kamailio-behind-nat/
http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-…
Cheers.
2016-05-06 09:59 tarihinde 王欣鑫 [via SIP
Router] yazdı:
Hello,ycaner!
Now I use a real IMS user data to register this user(for example
18616278715) to China Unicom's IMS through kamailio.
I can register successsfully, and can make call using this number,
the callee can pick up phone. However
neither party can hear counter part.
So, this should be rtp problem. And I captured a .pcap file on
port 5060 like below. In this, I see in SDP information
it says 192.168.199.124, it's a private address. So maybe the
callee can't setup a circuit to this ip, and thus no media was
transfered?
Should I uncomment RTPPROXY_ADDRESS, as in my pcscf.cfg there are
two lines. If I should do this, what ip and port should I use to
replace "udp:10.1.2.186:22222 udp:10.1.27.217:22222" ?
# IP-Adress(es) of the RTP-Proxy
##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222
udp:10.1.27.217:22222"
##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222"
This is the .pcap file captured on port 5060.
5060.pcap
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10:18 a.m.
Hello ycaner!
Thank you very much!
Now, what I think is maybe I should setup a rtpproxy server first, and then
modify some configuration in pcscf.cfg and kamailio.cfg.
But I will do a study into the reference you give me first.
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1:14 p.m.
Hello ycaner!
I download rtpproxy-2.0-RC1.tar.gz from
https://github.com/winer632/rtpproxy/releases/tag/V2.0-RC1
and did the following command:
tar -xzvf rtpproxy-2.0-RC1.tar.gz
cd rtpproxy-2.0-RC1
./configure
make all
make install
Then I start rtpproxy with this command:
rtpproxy -l 120.76.118.128 -s udp:localhost:7722
Then I edit kamailio.cfg in PCSCF module like this(added this
line*loadmodule "rtpproxy"* and *modparam("rtpproxy",
"rtpproxy_sock",
"udp:127.0.0.1:7722") * )
#!ifdef WITH_NAT
loadmodule "path"
loadmodule "rtpengine"
*loadmodule "rtpproxy"*
loadmodule "nathelper"
#!endif
#!ifdef WITH_NAT
# ----- rtpproxy-ng params -----
#modparam("rtpengine", "rtpengine_sock", RTPPROXY_ADDRESS)
#modparam("path", "use_received", 1)
*modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:7722")*
#!endif
Next, I register a real number(18616278715) in China unicom's IMS , this
step was success.Then call another real number 18019398639, no phone
ringing, but failed with "Status-Line: SIP/2.0 407 Proxy Authentication
Required". Before I add rtpproxy, there is no 407 error.
5060_unicom_407_error.pcap
<http://sip-router.1086192.n5.nabble.com/file/n148198/5060_unicom_407_error.pcap>
Before I add rtpproxy, I can hear the callee phone ringing, and the SIP
route is like this:
5060_unicom.pcap
<http://sip-router.1086192.n5.nabble.com/file/n148198/5060_unicom.pcap>
Am I doing wrong with rtpproxy? Maybe the problem is not here?
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2:07 p.m.
hello;
you can start rtpproxy like "rtpproxy -l 120.76.118.128 -s
udp:127.0.0.1:7722 -F -m 10000 -M 20000 -d DBUG:LOG_DAEMON" and
check message log.
After authantication , test only INVITE packets with
nat_uac_client(8) , rtpproxy_manage can handle INVITE packet with
"co" flags ( rtpproxy_manage("co") ) as specified on docs.
you can check 200 OK packet with t_on_reply packets. the same logic
can you handle sdp packets.
Cheers
2016-05-06 13:14 tarihinde 王欣鑫 [via SIP
Router] yazdı:
Hello ycaner!
I download rtpproxy-2.0-RC1.tar.gz from
https://github.com/winer632/rtpproxy/releases/tag/V2.0-RC1
and did the following command:
tar -xzvf rtpproxy-2.0-RC1.tar.gz
cd rtpproxy-2.0-RC1
./configure
make all
make install
Then I start rtpproxy with this command:
rtpproxy -l 120.76.118.128 -s udp:localhost:7722
Then I edit kamailio.cfg in PCSCF module like this(added this lineloadmodule
"rtpproxy" and modparam("rtpproxy",
"rtpproxy_sock",
"udp:127.0.0.1:7722") )
#!ifdef WITH_NAT
loadmodule "path"
loadmodule "rtpengine"
loadmodule "rtpproxy"
loadmodule "nathelper"
#!endif
#!ifdef WITH_NAT
# ----- rtpproxy-ng params -----
#modparam("rtpengine", "rtpengine_sock", RTPPROXY_ADDRESS)
#modparam("path", "use_received", 1)
modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:7722")
#!endif
Next, I register a real number(18616278715) in China unicom's IMS
, this step was success.Then call another real number 18019398639,
no phone ringing, but failed with "Status-Line: SIP/2.0 407 Proxy
Authentication Required". Before I add rtpproxy, there is no 407
error.
5060_unicom_407_error.pcap
Before I add rtpproxy, I can hear the callee phone ringing, and
the SIP route is like this:
5060_unicom.pcap
Am I doing wrong with rtpproxy? Maybe the problem is not here?
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