Thierry Luo:
我对 sip 各 rfc protocol 都很熟的。 但只做兼职工作。适合独立完成项目。
不知您的工作需求是什么？
George Wu
在 2015-09-28 14:22:34，"Thierry Luo" 写道：
很报歉通过这个邮件列表打扰大家。
我们是一个创业团队，刚刚拿到投资，正在组建核心团队，现需要一位在OpenSIPS方面有经验的C/C++开发工程师加盟，待遇优厚，期权可谈，职位在北京。请有意者速与luoyongh...@nane.cn联系。谢谢！
打
in.
About your manual UDP defrag - I do not think it is a good it - the UDP stack
must to that for you, you cannot simulate it on top of datagram.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.10.2014 14:05, george wu wrote:
Bogdan-Andrei:
7.10.2014 02:44, george wu wrote:
Hi, guys:
How can I adjust the global tcp keep alive time? what's the default in opensips?
I want it to close as soon as possible so that tcp deployment can scale.
George Wu
___
Users mailing list
Users@li
Hi, guys:
How can I adjust the global tcp keep alive time? what's the default in opensips?
I want it to close as soon as possible so that tcp deployment can scale.
George Wu
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.or
?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 15.10.2014 19:33, george wu wrote:
Is it because the packet MTU is too big? It can't handle size bigger than 1472
bytes?
The last line looks strange to me.
a=candidate:2 2 UDP 1694498814 124.1
efinitely shows linphone not answering to the INVITE handled via
mediaproxy (INVITE goes to linphone but nothing back).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 15.10.2014 17:32, george wu wrote:
//part 2
103.24.228.158.5060 > 124.193
Hi, Bogdan:
Is this a bug? When the server relay invite message to the callee in the last
line:
a=candidate:2 2 UDP 1694498814 124.193.138.210 5060 typ srflx raddr
192.168[|sip]
At 2014-10-15 22:31:41, "george wu" wrote:
Hi, Bogdan:
The following is the tcpdump with med
//part 2
103.24.228.158.5060 > 124.193.138.210.6001: SIP, length: 1472
INVITE sip:test2@192.168.1.3:5080 SIP/2.0
Record-Route:
Via: SIP/2.0/UDP 103.24.228.158:5060;branch=z9hG4bK57b9.3415f0e2.0
Via: SIP/2.0/UDP
192.168.1.3:5070;received=124.193.138.210;branch=z9hG4bK.FohJ-P
Hi, Bogdan:
The following is the tcpdump with mediaproxy on.
I can't see any problem. That means it is linphone problem which can't
understand ice?
Actually as I said before, if I use tcp/tls, ice actually works on linphone.
What should I do more to test it?
George Wu
that I can get more interesting result?
George Wu
[root@localhost ~]# tcpdump -nn -i eth2 'port 5060'
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on eth2, link-type EN10MB (Ethernet), capture size 65535 bytes
21:43:18.305220 IP 103.24.22
phone gets and prints out all the communication
which is correct.
George Wu
At 2014-10-15 20:46:02, "Bogdan-Andrei Iancu" wrote:
Hi George,
Not sure if a media relay process has anything to do with the ability to send
traffic to an UAC - do you actually see with ngrep/tcpdump
phone gets and prints out all the communication
which is correct.
George Wu
At 2014-10-15 20:46:02, "Bogdan-Andrei Iancu" wrote:
Hi George,
Not sure if a media relay process has anything to do with the ability to send
traffic to an UAC - do you actually see with ngrep/tcpdump
Can anybody share your mediaproxy configuration?
I am using mediaproxy to work with ice.
I modify the script from rtpproxy. Finally it turns out
it breaks some invite relay logic.
The route logic configuration is very hard. The original rtpproxy is generated
from menuconfig.
Since there is no op
g("incoming reply\n");
}
failure_route[missed_call] {
if (t_was_cancelled()) {
exit;
}
# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status("3[0-9][0-9]")) {
##t_reply("404"
My experience is for two uac (linphone) behind a firewall,
tcp/tls will always work.
udp will never work.
for both tcp/udp, my uac will send keep alive every 10 seconds.
I don't understand what makes those difference.
Can any one share your experience?
Geor
do i need nathelper if my client keeps pinging all the time?
What's the use of those two method? I think we should always use
5060 or 5061, otherwise how can you reach the client behind nat?
natping_socket (string)
force_socket (string)
Thanks.
Geor
Thank you very much, the mediaproxy works with ice quite nicely.
George Wu
At 2014-10-12 06:17:47, "Adrian Georgescu" wrote:
Try MediaProxy, it handles ICE negotiation properly.
Adrian
On 11 Oct 2014, at 10:26, george wu wrote:
When I use nathelper/rtpproxy, ice does not
When I use nathelper/rtpproxy, ice does not work well with it.
nathelper will rewrite the sdp media part which my ice client is not happy.
Then it will reply a=ice-mismatch. finally it will use rtpproxy to relay the
media.
the invite:
m=audio 36580 RTP/AVP 124 120 111 110 0 8 101
a=candidate
I think my client never send stun message to the server.
the client find its public ip from the OK VIA header:
Via: SIP/2.0/UDP
192.168.1.3:5070;received=192.168.1.3;branch=z9hG4bK.Klryaqrls;rport=5070
Am I right?
At 2014-10-10 23:36:18, "george wu" wrote:
Hi, all:
I can'
Hi, all:
I can't see any debug message for stun on the opensips server side.
I have enabled debug and set up stun as below
loadmodule "stun.so"
modparam("stun", "primary_ip", "192.168.1.3")
modparam("stun","alternate_ip","192.168.122.1")
From the client linphone debug message, I can see
ortp-mes
This might be a very simple question.
I can't find anything is /var/log/*
Can somebody help me? Thanks in advance!
George
### Global Parameters #
debug=3
log_stderror=no
log_facility=LOG_LOCAL0
fork=yes
children=4
/* uncomment the following lines to enable debugging */
debug=6
#for
I am not sure if I should start another thread or not.
I have similar question.
The client is android linphone. it says it needs to send keep-alive message for
firewall traversal.
For tcp it only needs to send every 10 minutes while for udp it needs to send
every 10 seconds.
It is obviously tcp
I can make opensips works for tcp and udp.
But the tls does not work.
I test it with linphone.
But it complains:
Could not start tls transport on port 5161, maybe this port is already used.
I google it and get
http://linuxexchange.org/questions/3383/error-when-configuring-linphone-to-use-sip-tls-
I only have one ip. Is it possible to setup stun?
My sip client is linphone. It uses ice and it requires a stun server.
I believe the required stun feature is just self address fixing.
I don't think it require any 2 ip features.
so how to set up the stun server with only one public ip?
loadmodule
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