[SIPForum-discussion] Port numbers problem

[Roni Even]

Hi,

According to RFC3550 it is not allowed to multiplex audio and video on the
same port, you can multiplex two audio streams or two video streams. What
you described is that the receiver expect to get the audio and video in UDP
port 18990

Roni Even

 

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Venkatesh.cherukuri
Sent: Monday, April 11, 2011 7:43 AM
To: discussion at sipforum.org; yasin at kaplan.net
Subject: Re: [SIPForum-discussion] Port numbers problem

 

Hi All,

                

                In all cases this is not correct (i.e. RTP packets can be
sent from any port). This is because if the other node expects RTCP
communication then it start sending receiver reports  on RTP port + 1   port
and expects sender reports. We can use single port for sending both audio
and video  packets but the next port should be always ready to receive RTCP
packets.

For ex:-

                You can use 

                RTP->    m=audio 18990

                RTP->    m= video 18990

                RTCP-> (18991 for receiving rtcp reports)


                Or if you want to use 5060 for sending RTP then you should
receive RTCP on 5061

 

This is no mandatory in all sip nodes as some  sip nodes only communicate
RTP only.

 

Best Regards,

Venkatesh Cherukuri.

 

 

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Gopalakrishnan A.N
Sent: Thursday, March 10, 2011 4:02 PM
To: Mudugere, Satish
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Port numbers problem

 

But I think, if you configure 5060 as RTP port then it should not conflict
the SIP port rite? please correct if i am wrong.

On Tue, Mar 8, 2011 at 11:38 PM, Mudugere, Satish
<satish.mudugere at intel.com> wrote:

Your understanding is correct. In SDP, devices share their receiving ports
(sent port can be anything). What ever is mentioned in media line is to be
the target media port.

-S


-----Original Message-----
From: andre heller [mailto:andre.heller1 at gmx.de]
Sent: Tuesday, March 08, 2011 2:24 AM
To: Mudugere, Satish; discussion at sipforum.org
Subject: Re: RE: [SIPForum-discussion] Port numbers problem

Hi Sathish,
you mean to say that it is absolutely fine to send RTP packets using 5060 as
local port, to the remote port number which I have received from remote
party's SDP. And when I need to listen for RTP then I should be listening on
the port number which I have sent in my INVITE/SDP to the remote party.

Finally I can understand it does not really matter from which port I send my
RTP.

Correct me if I am wrong.

Regards

-------- Original-Nachricht --------
> Datum: Mon, 7 Mar 2011 19:08:07 -0700
> Von: "Mudugere, Satish" <satish.mudugere at intel.com>
> An: andre heller <andre.heller1 at gmx.de>, "discussion at sipforum.org"
<discussion at sipforum.org>
> Betreff: RE: [SIPForum-discussion] Port numbers problem

> While socket binding local port (sending RTP packets from local to remote)
> can be anything, but the remote port must be the one received in media
> line of SDP. In your example it is 18990.
>
> -S
>
> -----Original Message-----
> From: discussion-bounces at sipforum.org
> [mailto:discussion-bounces at sipforum.org] On Behalf Of andre heller
> Sent: Monday, March 07, 2011 10:47 AM
> To: discussion at sipforum.org
> Subject: [SIPForum-discussion] Port numbers problem
>
> Hi,
> I need some clarification regarding binding port numbers when sending RTP
> packets.
>
> I am using network sockets in developing my SIP Client. When I send my RTP
> packets to the SIP Server which port should I locally bind to i.e my
> client should be binded to which port number, should it be the SIP Port =
5060,
> or the port number which I have sent with my INVITE which is m=audio 18990
> RTP/AVP.
>
> Can somebody give more information regarding this.
>
> Regards
> --
> NEU: FreePhone - kostenlos mobil telefonieren und surfen!
> Jetzt informieren: http://www.gmx.net/de/go/freephone
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org

--
NEU: FreePhone - kostenlos mobil telefonieren und surfen!
Jetzt informieren: http://www.gmx.net/de/go/freephone

_______________________________________________
This is the SIP Forum discussion mailing list
TO UNSUBSCRIBE, or edit your delivery options, please visit
http://sipforum.org/mailman/listinfo/discussion
Post to the list at discussion at sipforum.org




-- 
Thank you  with regards,
Gopalakrishnan A.N.

VoIP call - sip:saigop at gtalk2voip.com <mailto:sip%3Asaigop at gtalk2voip.com> 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20110411/5b6d1e4d/attachment-0002.html>