[OpenSIPS-Users] Connection failing between two linphone clients

[Bogdan-Andrei Iancu]

Hi ,

But the ACK has as Route hdr the IP of your server, so the caller part 
should send the ACK to your proxy (and not to the RURI indication) - 
your opensips is the one that will do routing based on RURI and send to 
callee.

Where exactly is the ACK lost ? between caller and proxy or between 
proxy and callee ?

Regards,
Bogdan


Yuvraj Pasi wrote:
> Hi everyone,
> I have installed the opensips 1.6.2 on our network . it has private 
> address 192.168.1.1. We are able to register using this
> server from behind our network as well as from outside our network 
> using public ip address.
> The problem is I am not able to make a conection between two linphone 
> clients which uses sip proxy.
> when i dug in i found out that the problem is in actually the INVITE 
> packet being transferred between two clients.
> >From what i know that opensips is supposed to change the contact 
> information in the sdp packet. so it does but
> it changes the contact information of the incoming INVITE packet from 
> public IP address to private IP address i.e.
> 192.168.1.1. & the other client ends up sending ACK packet to this 
> address. & thus it never reaches the first client.
>
> Message send from a client behind our netwrk only
>
> ortp-message-Message sent: (to dest=192.168.1.1:5060 
> <http://192.168.1.1:5060>)
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.1;branch=z9hG4bKc3ad.cfe10033.0
> Via: SIP/2.0/UDP 
> 59.161.177.228:5060;received=59.161.177.228;rport=5060;branch=z9hG4bK1023899182
> Record-Route: <sip:192.168.1.1;lr=on>
> From: <sip:tuser1 at PRIVATE_IP>;tag=332199248
> To: <sip:tuser2 at PRIVATE_IP>;tag=1564457814
> Call-ID: 637708402
> CSeq: 21 INVITE
> Contact: <sip:tuser2 at 192.168.1.248:5060 
> <http://sip:tuser2@192.168.1.248:5060>>
> Content-Type: application/sdp
> User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
> Content-Length:   231
>
> v=0
> o=root 123456 654321 IN IP4 192.168.1.248
> s=A conversation
> c=IN IP4 192.168.1.248
> t=0 0
> m=audio 7078 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> m=video 0 RTP/AVP 0
>
>
> Message received by other client outside our network.
>
> ortp-message-Received message:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 
> 59.161.177.228:5060;received=59.161.177.228;rport=5060;branch=z9hG4bK1023899182
> Record-Route: <sip:192.168.1.1;lr=on>
> From: <sip:tuser1 at PRIVATE_IP>;tag=332199248
> To: <sip:tuser2 at PRIVATE_IP>;tag=1564457814
> Call-ID: 637708402
> CSeq: 21 INVITE
> Contact: <sip:tuser2 at 192.168.1.248:5060;nat=yes>
> Content-Type: application/sdp
> User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
> Content-Length: 249
> P-hint: onreply_route|force_rtp_proxy
> P-hint: Onreply-route - fixcontact
>
> v=0
> o=root 123456 654321 IN IP4 192.168.1.248
> s=A conversation
> c=IN IP4 192.168.1.1
> t=0 0
> m=audio 55924 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> m=video 0 RTP/AVP 0
> a=nortpproxy:yes
>
>
> & hence it send the ACK message as
>
> ortp-message-Message sent: (to dest=192.168.1.1:5060 
> <http://192.168.1.1:5060>)
> ACK sip:tuser2 at 192.168.1.248:5060;nat=yes SIP/2.0
> Via: SIP/2.0/UDP 59.161.177.228:5060;rport;branch=z9hG4bK2056355642
> Route: <sip:192.168.1.54;lr=on>
> From: <sip:tuser1 at PRIVATE_IP>;tag=332199248
> To: <sip:tuser2 at PRIVATE_IP>;tag=1564457814
> Call-ID: 637708402
> CSeq: 21 ACK
> Contact: <sip:tuser1 at 59.161.177.228:5060 
> <http://sip:tuser1@59.161.177.228:5060>>
> Proxy-Authorization: Digest username="tuser1", realm="PRIVATE_IP5", 
> nonce="4c0f700200000037926169c961edd9ac10258ccf5fa75912", 
> uri="sip:tuser2 at PRIVATE_IP"
> , response="00dab0a4a4a14d24a2a9a69daf5136ee", algorithm=MD5
> Max-Forwards: 70
> User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
> Content-Length: 0
>
> & therefore this ACK never reaches our server & the call gets dropped.
>
> The same scenario stands true even if both the clients are out side 
> our network. I have written the opensips.cfg file exactly as mentioned 
> in the book
> 'Building Telephony system using Opensips1.6' by flavio gnocalves.
>
> If somebody has faced this problem please guide me in the right direction.
> Any help will be appreciated.
>
> -- 
> Thanks & regards
> yuvraj pasi
> ------------------------------------------------------------------------
>
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>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro