7 May
2015
7 May
'15
10:01 a.m.
Hi Daniel
I have rtpproxy to perform media relaying.
The case where it just timeouts:
1. Client A behind NAT (wifi) calling Client B on 3G
Sent from Samsung Mobile
-------- Original message --------
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date:05/05/2015 12:52 (GMT+05:30)
To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] Kamailio proxy for far-end nat traversal
Hello,
do you have rtpproxy or rtpengine for relaying rtp packets? Kamailio is routing only sip
packets, you need the rtp relay application to help with media streams.
Cheers,
Daniel
On 01/05/15 13:15, rahul.ultimate wrote:
Hello
I need a small guidance on creating a light weight proxy which only forwards the msgs to
my sip server and also does supports symmetrical nated clients.
The way I have created the configuration is a slight modification of :
https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg
Problem is unless the sip clients use STUN media packets are not routed. And sometimes
even the signalling does not pass through.
Should i not use fix_nated_register and switch to fix _ nated_contact always ?
Because in all those cases where signalling does not pass through, I see RTO. My best
guess is kamailio trying to communicate with private ip.
Anything which I can try that you could suggest from the top of yourmind woyld be muc
appreciated.
Thanks
Rahul
Sent from Samsung Mobile
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
8 May
8 May
3:24 a.m.
New subject: Kamailio proxy for far-end nat traversal
Hello,
you can force rtp relaying always, if it is what you want -- just call
rtpproxy_manage() for all INVITE requests and their replies.
Otherwise you have to tune the nat_uac_test() parameter in order to
match what you want to relay.
For signaling relaying, be sure you do record_route() for calls.
Cheers,
Daniel
On 07/05/15 16:01, rahul.ultimate wrote:
Hi Daniel
I have rtpproxy to perform media relaying.
The case where it just timeouts:
1. Client A behind NAT (wifi) calling Client B on 3G
Sent from Samsung Mobile
-------- Original message --------
From: Daniel-Constantin Mierla
Date:05/05/2015 12:52 (GMT+05:30)
To: "Kamailio (SER) - Users Mailing List"
Subject: Re: [SR-Users] Kamailio proxy for far-end nat traversal
Hello,
do you have rtpproxy or rtpengine for relaying rtp packets? Kamailio
is routing only sip packets, you need the rtp relay application to
help with media streams.
Cheers,
Daniel
On 01/05/15 13:15, rahul.ultimate wrote:
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
Hello
I need a small guidance on creating a light weight proxy which only
forwards the msgs to my sip server and also does supports symmetrical
nated clients.
The way I have created the configuration is a slight modification of :
https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg
Problem is unless the sip clients use STUN media packets are not
routed. And sometimes even the signalling does not pass through.
Should i not use fix_nated_register and switch to fix _ nated_contact
always ?
Because in all those cases where signalling does not pass through, I
see RTO. My best guess is kamailio trying to communicate with private ip.
Anything which I can try that you could suggest from the top of
yourmind woyld be muc appreciated.
Thanks
Rahul
Sent from Samsung Mobile
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
19 May
19 May
1:41 p.m.
New subject: Kamailio proxy for far-end nat traversal
Hello Daniel,
Thanks for replying back !
And please accept my apologies for responding late. I tried modifying the
configuration file to do a far-end NAT traversal but this case (wifi to 3G)
is still not working.
Below is the trace of what is happening on my systems -
Registration of UAC Behind WiFi
*******************************
UAC ------> Proxy
-----------------
REGISTER sip:188.227.178.226 SIP/2.0
Via: SIP/2.0/UDP 59.178.140.180:5060;branch=z9hG4bK.MAq2ODF9T;rport
From: <sip:181085@188.227.178.226>;tag=BQR7JcwOi
To: sip:181085@188.227.178.226
CSeq: 25 REGISTER
Call-ID: GCOzYfxLqw
Max-Forwards: 70
Supported: replaces, outbound
Contact: <sip:181085@59.178.140.180
;+sip.instance="<urn:uuid:b368e9b2-d156-4377-8e5e-b1b6f293de45>"
Expires: 3600
User-Agent: LinphoneAndroid/3 (belle-sip/1.3.2)
Authorization: Digest realm="188.227.178.226",
nonce="555b738d000137e5140acc6ffeafac33d1aafa6094b3da16",
username="181085", uri="sip:188.227.178.226",
response="d622655dcae5c6861c05e71533cb445e"
Proxy ------> SIP Server
------------------------
REGISTER sip:188.227.178.226 SIP/2.0
Via: SIP/2.0/UDP 104.222.98.124:7878
;branch=z9hG4bKf61e.10e33f9242f9bab1186af8c0d860f197.0
Via: SIP/2.0/UDP 59.178.140.180:5060
;received=59.178.140.180;branch=z9hG4bK.MAq2ODF9T;rport=5060
From: <sip:181085@188.227.178.226>;tag=BQR7JcwOi
To: sip:181085@188.227.178.226
CSeq: 25 REGISTER
Call-ID: GCOzYfxLqw
Max-Forwards: 69
Supported: replaces, outbound
Contact: <sip:181085@59.178.140.180
;+sip.instance="<urn:uuid:b368e9b2-d156-4377-8e5e-b1b6f293de45>"
Expires: 3600
User-Agent: LinphoneAndroid/3 (belle-sip/1.3.2)
Authorization: Digest realm="188.227.178.226",
nonce="555b738d000137e5140acc6ffeafac33d1aafa6094b3da16",
username="181085", uri="sip:188.227.178.226",
response="d622655dcae5c6861c05e71533cb445e"
P-hint: outbound
SIP Server ------> Proxy
------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 104.222.98.124:7878
;received=104.222.98.124;rport=7878;branch=z9hG4bKf61e.10e33f9242f9bab1186af8c0d860f197.0
Via: SIP/2.0/UDP 59.178.140.180:5060
;received=59.178.140.180;branch=z9hG4bK.MAq2ODF9T;rport=5060
From: <sip:181085@188.227.178.226>;tag=BQR7JcwOi
To: sip:181085@188.227.178.226;tag=66565ae6872f3b1972fa74c2273140bf.91f7
CSeq: 25 REGISTER
Call-ID: GCOzYfxLqw
Contact: <sip:181085@59.178.140.180>;expires=120
Server: OpenSIPS (1.8.3-notls (x86_64/linux))
Content-Length: 0
Proxy ------> UAC
-----------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 59.178.140.180:5060
;received=59.178.140.180;branch=z9hG4bK.MAq2ODF9T;rport=5060
From: <sip:181085@188.227.178.226>;tag=BQR7JcwOi
To: sip:181085@188.227.178.226;tag=66565ae6872f3b1972fa74c2273140bf.91f7
CSeq: 25 REGISTER
Call-ID: GCOzYfxLqw
Contact: <sip:181085@59.178.140.180>;expires=120
Server: OpenSIPS (1.8.3-notls (x86_64/linux))
Content-Length: 0
Registration of UAC on 3G
*************************
UAC ------> Proxy
-----------------
REGISTER sip:188.227.178.226 SIP/2.0
Via: SIP/2.0/UDP 106.201.89.50:5060;branch=z9hG4bK.TZ8-sBaxi;rport
From: <sip:10185@188.227.178.226>;tag=bL0ESqY70
To: sip:10185@188.227.178.226
CSeq: 25 REGISTER
Call-ID: ei4dEgWhzV
Max-Forwards: 70
Supported: replaces, outbound
Contact: <sip:10185@106.201.89.50
;+sip.instance="<urn:uuid:61dd7345-d790-4146-8a00-7df4478193e6>"
Expires: 3600
User-Agent: LinphoneAndroid/3 (belle-sip/1.3.2)
Authorization: Digest realm="188.227.178.226",
nonce="555b7343000137493797b541c03d370fd95d1c5f1b634449",
username="10185",
uri="sip:188.227.178.226",
response="1e9f404568b1fbded560fafc8714db31"
Proxy ------> SIP Server
------------------------
REGISTER sip:188.227.178.226 SIP/2.0
Via: SIP/2.0/UDP 104.222.98.124:7878
;branch=z9hG4bKdf92.7d24b9515de3bb87a736bb7e40cd1923.0
Via: SIP/2.0/UDP 106.201.89.50:5060
;received=106.201.89.50;branch=z9hG4bK.TZ8-sBaxi;rport=5060
From: <sip:10185@188.227.178.226>;tag=bL0ESqY70
To: sip:10185@188.227.178.226
CSeq: 25 REGISTER
Call-ID: ei4dEgWhzV
Max-Forwards: 69
Supported: replaces, outbound
Contact: <sip:10185@106.201.89.50
;+sip.instance="<urn:uuid:61dd7345-d790-4146-8a00-7df4478193e6>"
Expires: 3600
User-Agent: LinphoneAndroid/3 (belle-sip/1.3.2)
Authorization: Digest realm="188.227.178.226",
nonce="555b7343000137493797b541c03d370fd95d1c5f1b634449",
username="10185",
uri="sip:188.227.178.226",
response="1e9f404568b1fbded560fafc8714db31"
P-hint: outbound
SIP Server ------> Proxy
------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 104.222.98.124:7878
;received=104.222.98.124;rport=7878;branch=z9hG4bKaf92.109b3a2c233dcc8d0a0adafe734e77d5.0
Via: SIP/2.0/UDP 106.201.89.50:5060
;received=106.201.89.50;branch=z9hG4bK.u6mLnvNhh;rport=5060
From: <sip:10185@188.227.178.226>;tag=bL0ESqY70
To: sip:10185@188.227.178.226;tag=66565ae6872f3b1972fa74c2273140bf.caca
CSeq: 26 REGISTER
Call-ID: ei4dEgWhzV
Contact: <sip:10185@106.201.89.50>;expires=120
Server: OpenSIPS (1.8.3-notls (x86_64/linux))
Content-Length: 0
Proxy ------> UAC
-----------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 106.201.89.50:5060
;received=106.201.89.50;branch=z9hG4bK.u6mLnvNhh;rport=5060
From: <sip:10185@188.227.178.226>;tag=bL0ESqY70
To: sip:10185@188.227.178.226;tag=66565ae6872f3b1972fa74c2273140bf.caca
CSeq: 26 REGISTER
Call-ID: ei4dEgWhzV
Contact: <sip:10185@106.201.89.50>;expires=120
Server: OpenSIPS (1.8.3-notls (x86_64/linux))
Content-Length: 0
Calling from UAa (wifi) to UAb (3G)
***********************************
UAa ------> Proxy
-----------------
INVITE sip:10185@188.227.178.226 SIP/2.0
Via: SIP/2.0/UDP 59.178.140.180:5060;branch=z9hG4bK.dii5ugxwb;rport
From: <sip:181085@188.227.178.226>;tag=sjR6XkprZ
To: sip:10185@188.227.178.226
CSeq: 20 INVITE
Call-ID: 5kP-E1uHL1
Max-Forwards: 70
Route: <sip:104.222.98.124:7878;lr>
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 421
Contact: <sip:181085@59.178.140.180
;+sip.instance="<urn:uuid:b368e9b2-d156-4377-8e5e-b1b6f293de45>"
User-Agent: LinphoneAndroid/3 (belle-sip/1.3.2)
v=0
o=181085 2051 698 IN IP4 59.178.140.180
s=Talk
c=IN IP4 59.178.140.180
b=AS:380
t=0 0
m=audio 11695 RTP/AVP 124 120 111 110 0 8 101
a=rtpmap:124 opus/48000/2
a=fmtp:124 useinbandfec=1; stereo=0; sprop-stereo=0
a=rtpmap:120 SILK/16000
a=rtpmap:111 speex/16000
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:7077 IN IP4 192.168.1.2
Proxy ------> SIP Server
------------------------
INVITE sip:10185@188.227.178.226 SIP/2.0
Record-Route: <sip:104.222.98.124:7878;lr=on;nat=yes>
Via: SIP/2.0/UDP 104.222.98.124:7878
;branch=z9hG4bKae74.d03682b95a7f012aaa8cc9e9bc9b14f5.0
Via: SIP/2.0/UDP 59.178.140.180:5060
;received=59.178.140.180;branch=z9hG4bK.dii5ugxwb;rport=5060
From: <sip:181085@188.227.178.226>;tag=sjR6XkprZ
To: sip:10185@188.227.178.226
CSeq: 20 INVITE
Call-ID: 5kP-E1uHL1
Max-Forwards: 69
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 421
Contact: <sip:181085@59.178.140.180
;+sip.instance="<urn:uuid:b368e9b2-d156-4377-8e5e-b1b6f293de45>"
User-Agent: LinphoneAndroid/3 (belle-sip/1.3.2)
Path: <sip:104.222.98.124:7878;lr;received=sip:59.178.140.180:5060>
P-hint: outbound
v=0
o=181085 2051 698 IN IP4 104.222.98.124
s=Talk
c=IN IP4 104.222.98.124
b=AS:380
t=0 0
m=audio 19880 RTP/AVP 124 120 111 110 0 8 101
a=rtpmap:124 opus/48000/2
a=fmtp:124 useinbandfec=1; stereo=0; sprop-stereo=0
a=rtpmap:120 SILK/16000
a=rtpmap:111 speex/16000
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:19881
a=nortpproxy:yes
SIP Server ------> Proxy
------------------------
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 104.222.98.124:7878
;received=104.222.98.124;rport=7878;branch=z9hG4bKbe74.a7aaf30744452fb8d6e4a007c5454b22.0
Via: SIP/2.0/UDP 59.178.140.180:5060
;received=59.178.140.180;branch=z9hG4bK.xKUkXp0dg;rport=5060
From: <sip:181085@188.227.178.226>;tag=sjR6XkprZ
To: sip:10185@188.227.178.226
CSeq: 21 INVITE
Call-ID: 5kP-E1uHL1
Server: OpenSIPS (1.8.3-notls (x86_64/linux))
Content-Length: 0
SIP Server ------> UAb
----------------------
INVITE sip:10185@106.201.89.50 SIP/2.0
Record-Route: <sip:188.227.178.226;lr;did=b63.3b918bd2>
Record-Route: <sip:104.222.98.124:7878;lr=on;nat=yes>
Via: SIP/2.0/UDP 188.227.178.226:5060;branch=z9hG4bKbe74.e083e367.0
Via: SIP/2.0/UDP 104.222.98.124:7878
;rport=7878;received=104.222.98.124;branch=z9hG4bKbe74.a7aaf30744452fb8d6e4a007c5454b22.0
Via: SIP/2.0/UDP 59.178.140.180:5060
;received=59.178.140.180;branch=z9hG4bK.xKUkXp0dg;rport=5060
From: <sip:181085@188.227.178.226>;tag=sjR6XkprZ
To: sip:10185@188.227.178.226
CSeq: 21 INVITE
Call-ID: 5kP-E1uHL1
Max-Forwards: 68
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 421
Contact: <sip:181085@59.178.140.180
;+sip.instance="<urn:uuid:b368e9b2-d156-4377-8e5e-b1b6f293de45>"
User-Agent: LinphoneAndroid/3 (belle-sip/1.3.2)
Path: <sip:104.222.98.124:7878;lr;received=sip:59.178.140.180:5060>
P-hint: outbound
v=0
o=181085 2051 698 IN IP4 104.222.98.124
s=Talk
c=IN IP4 104.222.98.124
b=AS:380
t=0 0
m=audio 10326 RTP/AVP 124 120 111 110 0 8 101
a=rtpmap:124 opus/48000/2
a=fmtp:124 useinbandfec=1; stereo=0; sprop-stereo=0
a=rtpmap:120 SILK/16000
a=rtpmap:111 speex/16000
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:10327
a=nortpproxy:yes
Finally,
SIP Server ------> Proxy
------------------------
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 104.222.98.124:7878
;received=104.222.98.124;rport=7878;branch=z9hG4bKbe74.a7aaf30744452fb8d6e4a007c5454b22.0
Via: SIP/2.0/UDP 59.178.140.180:5060
;received=59.178.140.180;branch=z9hG4bK.xKUkXp0dg;rport=5060
From: <sip:181085@188.227.178.226>;tag=sjR6XkprZ
To: sip:10185@188.227.178.226;tag=9da0519e96ad88f8de3fd8d2c9042128-3485
CSeq: 21 INVITE
Call-ID: 5kP-E1uHL1
Server: OpenSIPS (1.8.3-notls (x86_64/linux))
Content-Length: 0
Kindly guide me to resolve this part, I am attaching the kamailio.cfg file
herewith.
Thanks
Rahul
On Fri, May 8, 2015 at 12:54 PM, Daniel-Constantin Mierla <miconda(a)gmail.com
wrote:
Hello,
you can force rtp relaying always, if it is what you want -- just call
rtpproxy_manage() for all INVITE requests and their replies.
Otherwise you have to tune the nat_uac_test() parameter in order to match
what you want to relay.
For signaling relaying, be sure you do record_route() for calls.
Cheers,
Daniel
On 07/05/15 16:01, rahul.ultimate wrote:
Hi Daniel
I have rtpproxy to perform media relaying.
The case where it just timeouts:
1. Client A behind NAT (wifi) calling Client B on 3G
Sent from Samsung Mobile
-------- Original message --------
From: Daniel-Constantin Mierla
Date:05/05/2015 12:52 (GMT+05:30)
To: "Kamailio (SER) - Users Mailing List"
Subject: Re: [SR-Users] Kamailio proxy for far-end nat traversal
Hello,
do you have rtpproxy or rtpengine for relaying rtp packets? Kamailio is
routing only sip packets, you need the rtp relay application to help with
media streams.
Cheers,
Daniel
On 01/05/15 13:15, rahul.ultimate wrote:
Hello
I need a small guidance on creating a light weight proxy which only
forwards the msgs to my sip server and also does supports symmetrical nated
clients.
The way I have created the configuration is a slight modification of :
https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg
Problem is unless the sip clients use STUN media packets are not routed.
And sometimes even the signalling does not pass through.
Should i not use fix_nated_register and switch to fix _ nated_contact
always ?
Because in all those cases where signalling does not pass through, I see
RTO. My best guess is kamailio trying to communicate with private ip.
Anything which I can try that you could suggest from the top of yourmind
woyld be muc appreciated.
Thanks
Rahul
Sent from Samsung Mobile
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
--
Warm Regds.
MathuRahul
20 May
20 May
5:13 a.m.
New subject: Kamailio proxy for far-end nat traversal
Looks like UAb (3G) do not receive the INVITE (or makes no answer for some
reason).
Can you check if UAb is receiving the INVITE?
LinphoneAndroid that you use as a User-Agent on UAb has to log something.
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Kamailio-proxy-for-far-end-nat-trav…
Sent from the Users mailing list archive at Nabble.com.
9:56 a.m.
New subject: Kamailio proxy for far-end nat traversal
Hello Vasiliy,
Thanks for replying.
Not sure why Linphone-Android won't receive INVITE, since it is responding
well to the keep-alive OPTIONS messages from proxy.
Both, proxy and SIP server are sending packets to UAb on which UAb is
apparently responding to only proxy.
Is this a genuine flaw in Linphone OR some flaw in my configuration of
proxy ?
On Wed, May 20, 2015 at 2:43 PM, Vasiliy Ganchev <vasiliy.ganchev(a)wildix.com
wrote:
Looks like UAb (3G) do not receive the INVITE (or
makes no answer for some
reason).
Can you check if UAb is receiving the INVITE?
LinphoneAndroid that you use as a User-Agent on UAb has to log something.
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Kamailio-proxy-for-far-end-nat-trav…
Sent from the Users mailing list archive at Nabble.com.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Warm Regds.
MathuRahul
21 May
21 May
12:52 p.m.
New subject: Kamailio proxy for far-end nat traversal
Hello Daniel, Vasiliy
Did you get a chance to look at it ?
I tried the same experiment with CSipSimple + Video plugin but results are
unfortunately the same :(
So does it comes down to the configuration of the proxy ? Or something else
?
Really need some guidance here.
Thanks!
On Wed, May 20, 2015 at 7:26 PM, Rahul MathuR <rahul.ultimate(a)gmail.com>
wrote:
Hello Vasiliy,
Thanks for replying.
Not sure why Linphone-Android won't receive INVITE, since it is responding
well to the keep-alive OPTIONS messages from proxy.
Both, proxy and SIP server are sending packets to UAb on which UAb is
apparently responding to only proxy.
Is this a genuine flaw in Linphone OR some flaw in my configuration of
proxy ?
On Wed, May 20, 2015 at 2:43 PM, Vasiliy Ganchev <
vasiliy.ganchev(a)wildix.com> wrote:
--
Warm Regds.
MathuRahul
Looks like UAb (3G) do not receive the INVITE (or
makes no answer for some
reason).
Can you check if UAb is receiving the INVITE?
LinphoneAndroid that you use as a User-Agent on UAb has to log something.
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MathuRahul
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5 comments
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participants (4)
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Daniel-Constantin Mierla -
Rahul MathuR -
rahul.ultimate
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Vasiliy Ganchev