SIP typically doesn't work that way. while it's possible most(almost all)
SIP clients are written to expect a text based SIP message on the UDP port
on which it listens, while the RTP handler expects UDP message with a binary
payload (with a valid RTP header). To avoid having the UDP message
handler figure out whether a RTP formatted or a SIP formatted message has
arrived separate UDP ports are used. It just makes thing easier.
Regards
Peter
On Wed, Feb 3, 2010 at 10:09 PM, <[email protected]> wrote:
> Hi,
>
> As the transport used is UDP hence I think we can use same port for both
> signalling and audio.
>
> Regards
> Sunil Verma
>
> -----Original Message-----
> From: [email protected]
> [mailto:[email protected]] On Behalf Of
> Pavesi, Valdemar (NSN - US/Boca Raton)
> Sent: Wednesday, February 03, 2010 9:51 PM
> To: ext Pranab Bohra; Nahum Nir
> Cc: [email protected]
> Subject: Re: [Sip-implementors] What is wrong with the following
> INVITE???
>
> Hi,
>
> But it should not be the problem , the endpoint-b don't nothing about
> the same port for sip/rtp.
>
> You must see a RTP-stream coming to your device. (endpoint-a)
>
>
> Regards!
> Valdemar
>
>
>
> -----Original Message-----
> From: [email protected]
> [mailto:[email protected]] On Behalf Of ext
> Pranab Bohra
> Sent: Wednesday, February 03, 2010 11:04 AM
> To: Nahum Nir
> Cc: [email protected]
> Subject: Re: [Sip-implementors] What is wrong with the following
> INVITE???
>
> Hi,
>
> You are using port 44074 for both SIP signalling and RTP, generally
> media and signalling are handled on different ports.
>
> Cheers,
> Pranab
>
> On Wed, Feb 3, 2010 at 3:15 PM, Nahum Nir <[email protected]>
> wrote:
> > Hi All,
> >
> > I'm trying to make an INVITE so that the incoming audio will be
> received on
> > 89.139.126.116:44074.
> >
> > I get Session In Progress and OK but no incoming audio.
> > What is wrong???
> >
> > Thanks,
> > Nahum
> >
> > INVITE
> sip:[email protected]<sip%3a%[email protected]>
> <sip%3a%[email protected].
> theiptele.com>SIP/2.0
> > Via: SIP/2.0/UDP 89.139.126.116:44074
> > ;rport;branch=z9hG4bkxu9h74udx9n9jhm4fp35vfurr7ao7x8nu
> > Max-Forwards:
> > 70
> >
> > From:
> sip:[email protected]<sip%[email protected]>
> <sip%[email protected] <sip%[email protected]>
> eiptele.com>;tag=74tosdtd1fjmj03nxkdqjicczu7tw0
> >
> > To:
> sip:[email protected]<sip%3a%[email protected]>
> <sip%3a%[email protected].
> theiptele.com>
> >
> > Call-ID:
> > bcu20hjb9solhzl5q70ymf22hxhg10xm
> >
> > CSeq: 1001
> > INVITE
> >
> > Contact: <sip:[email protected]:44074;transport=UDP>
> >
> > Authorization: Digest username="0001001103826", realm="88.215.60.14",
> > nonce="8191cd28e8d275daaf2b6b441ac804534b69414e", uri="
> > sip:[email protected] <sip%3a%[email protected]> <
> sip%3a%[email protected]<sip%253a%[email protected]>
> >",
> > response="63fa61f9eae18f1e0c4bd1be37c30406",
> > algorithm=md5
> >
> > Content-Type:
> > application/sdp
> >
> > Content-Length:
> > 176
> >
> >
> > v=0
> > o=- 1265189596 1265189596 IN IP4 89.139.126.116 s=nsip server version
> > 1.0 c=IN IP4 89.139.126.116 t=0 0 m=audio 44074 RTP/AVP 8 a=sendrecv
> > a=rtpmap:8 PCMA/8000
> > _______________________________________________
> > Sip-implementors mailing list
> > [email protected]
> > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
> >
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