Hi Attila,
thanks for your reply. I am sure that I know this is how it should work but
when I try to use the local RTP port (22456), I could not send anything to
the server. On the other hand when I change the local RTP port number to
5060, I can see the RTP packets on my network sniffer like this

1.2.3.4 ---> 3.4.5.6 RTP PT=ITU-T G.711 PCMA, SSRC=0xDF53, Seq = 45,
Time=12960 (Client to Server)
1.2.3.4 ---> 3.4.5.6 RTP PT=ITU-T G.711 PCMA, SSRC=0xDF53, Seq = 46,
Time=13120.
3.4.5.6 ---> 1.2.3.4 RTP PT=ITU-T G.711 PCMA, SSRC=0xF231, Seq = 3457,
Time=160(Server to Client)
1.2.3.4 ---> 3.4.5.6 RTP PT=ITU-T G.711 PCMA, SSRC=0xDF53, Seq = 47,
Time=13280.
3.4.5.6 ---> 1.2.3.4 RTP PT=ITU-T G.711 PCMA, SSRC=0xF231, Seq = 3458,
Time=320

and so on....

that is what I dont understand, how this is happening. So in general am I
making some weird mistake.

Regards




On Mon, Mar 7, 2011 at 6:47 PM, Attila Sipos <[email protected]>wrote:

> >> is this normal to send RTP packets from the local port = *5060(SIP
> PORT)*
> >> with remote port = *18564 *and receive RTP packets from SIP Server on
> local
> >> port = *22456 * with remote port = *18564.
>
> You are confusing 2 separate things:
> 1. there is a SIP signalling port 5060 - this is used only to receive
> SIP messages
> 2. there is a RTP port 22456
>
> Your RTP packets should use the local RTP port (22456) as the RTP port.
>
>
> 1. SIP and RTP are separate
> 2. SIP and RTP use separate ports
> 3. do not mix SIP and RTP ports - in your example you used the SIP port
> as the local RTP port - this will not work!
>
> I hope this helps
>
> Regards
>
> Attila
>
>
>
>
> -----Original Message-----
> From: [email protected]
> [mailto:[email protected]] On Behalf Of
> Siga
> Sent: 07 March 2011 17:32
> To: [email protected]
> Subject: [Sip-implementors] Ports for sending and receiving RTP packets!
>
> Hi,
> I am facing rather a strange problem with the ports for sending and
> receiving RTP packets.
> This is the second time I am posting my question.
>
> Initially I am able to set up a SIP Client successfully
>
> SIP CLIENT                            SIP SERVER
>  (1.2.3.4)                                 (3.4.5.6)
>
>
>          INVITE: 1.2.3.4:5060
>          -------------------------------------------->
>
>            100 Trying: 3.4.5.6
>        <---------------------------------------------
>
>
>             200 OK: 3.4.5.6
>          <--------------------------------------------
>
>
>                  ACK: 1.2.3.4:5060
>           -------------------------------------------->
>
>
>              RTP Packets
>        -------------------------------------------->
>
> Here comes the problem: my SIP Client INVITE looks like this
>
>                               "INVITE
> sip:[email protected]/2.0\r\n"
>                                "Via:SIP/2.0/UDP
> 192.168.x.000:5060;branch=z9hG4bKdg18\r\n"
>                                 "Max-Forwards: 70\r\n"
>                                 "To: server <sip:[email protected]>\r\n"
>                                 "From: User<sip:[email protected]>;
> tag = 76341\r\n"
>                                 "Call-ID:
> [email protected]@192.168.x.x\r\n"
>                                 "CSeq: 1 INVITE\r\n"
>                                 "Contact:
> <sip:[email protected]>\r\n"
>                                 "Content-Type: application/sdp\r\n"
>                                 "Content-Length:142\r\n";
>                                 "\r\n"
>                                 "v=0\r\n"
>                                 "o=User 53655765 2353687637 IN IP4
> 192.168.x.000\r\n"
>                                 "s=-\r\n"
>                                 "c=IN IP4 192.168.x.000\r\n"
>                                 "t=0 0\r\n"
>                                 "*m=audio 22456 *RTP/AVP 0 8\r\n"
>                                 "m=video 51372 RTP/AVP 98 49\r\n"
>                                 "a=rtpmap:0 PCMA/8000\r\n"
>                                 "a=rtpmap:98 H263-1999/90000\r\n";
>
> that is here I am mentioning that I am available on the port =  *22456 ,
> *for receiving RTP packets.
>
> On the other hand from the *200 OK SDP* which I get from the *SIP
> Server* I am parsing the audio port number = *18564 *which I use to send
> the audio RTP packets from the Client side.
>
> Until here everything is fine, but when I try to bind my client to the
> local
> address(1.2.3.4) and port number, there are two scenarios
>
> 1. If I bind my client to the local port = *22456, *this is how I am
> doing* **Local.sin_port = 22456**;  *When I have done like this* *I cant
> send anything to the SIP Server, after time out SIP Server sends BYE to
> the SIP Client.
>
> 2. If I bind my client to the local port = *5060 (SIP PORT),
> **Local.sin_port = 5060**;  *When I have done like this* *I can send RTP
> packets to the SIP Server and there is a constant flow of packets which
> I can capture and view using Wireshark.
>
> For both the above cases, I am declaring the remote port for the Server
> in this manner* remote.sin_port = **18564 *
>
> is this normal to send RTP packets from the local port = *5060(SIP
> PORT)* with remote port = *18564 *and receive RTP packets from SIP
> Server on local port = *22456 * with remote port = *18564.
>
> *I am really confused here*. *If I am doing something wrong then please
> correct me*.
>
> *If you need any further information then please let me know.
>
> Regards*
> *
> _______________________________________________
> Sip-implementors mailing list
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