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@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

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@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-trave... Sent from the Users mailing list archive at Nabble.com.

---

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users