[SIPForum-discussion] SDP Message in SIP problem. Please Help.

[Medve, Steve]

"SIP" uses ports 5060 and higher while RTP has no Fixed range of ports
to use.  This may cause a conflict but haven't been privy to see that
yet.  I know the port for RTP should be even numbered and the
corresponding RTCP should be odd numbered ports.  

Pulled from some notes I had:

To setup an RTP session, an application defines a pair of destination
ports (an IP address with a pair of ports for RTP and RTCP). In a
multimedia session, each media stream is carried in a separate RTP
session, with its own RTCP packets reporting the reception quality for
that session. For example, audio and video would travel in separate RTP
session, enabling a receiver to select whether or not to receive a
particular stream.[10] An RTP port should be even and the RTCP port
should be the next higher port number if possible. Deviations from this
rule can be signaled via RTP session descriptions in other protocols
(SDP). RTP and RTCP typically use unprivileged UDP ports (1024 to
65535),[11] but may use other transport protocols (most notably, SCTP
and DCCP) as well, as the protocol design is transport independent.

Voice over Internet Protocol (VoIP) systems most often use the Session
Description Protocol (SDP)[12] to define RTP sessions and negotiate the
parameters involved with other peers. The Real Time Streaming Protocol
(RTSP) may be also be used to setup and control media session on remote
media servers.

Hope it helps. 

Steven A. Medve 
REDCOM Laboratories, Inc.
One Redcom Center
Victor, NY 14564-0995
585-924-6552 - Office
315-372-3045 - Mobile
smedve at redcom.com
www.redcom.com



-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Kristine Regis
Sent: Monday, July 27, 2009 10:34 PM
To: Anthony Orlando
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] SDP Message in SIP problem. Please
Help.

Hi,

Thanks for the reply but can you give me more details why you would not 
use port 5060 for the rtp. I am still a beginner in this field that is 
why I need help regarding this. Thank you.

-Kristine

Anthony Orlando wrote:
> I wouldn't use port 5060 for the rtp. 
>
> Sent from my wonderful iPhone!
>
> On Jul 25, 2009, at 3:38, Kristine Regis <kristine at racequeen.ph>
wrote:
>
> Hi, I hope anyone could help me figure out what's the problem here.
>
> I am currently developing a SIP Call. The flow of the call from INVITE

> to BYE is correct, as I assume since the server accepted it - the
server 
> is OpenSIPs. My SDP is as follows:
>
> v=0
> o=root 3457409648 3457409648 IN IP4 172.16.100.21
> s=-
> c=IN IP4 172.16.100.21
> t=0 0
> m=audio 5060 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
>
>
> I have this SDP and the voice streaming cannot be done. The caller and

> the call cannot here each other. I don't know what is the problem here

> or if the SDP above is wrong.
>
> I hope you could help me solve this problem. Thank you.
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>       
>
>   

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