Hi Siga, Yes, it is absolutely fine if you send RTP from 5060. But just a suggestion try NOT to use standard SIP port. Maybe you can use a port range of 50000 to 54000 for RTP keeping in mind that an RTP port should be even and the RTCP port is the next higher odd port number.
>Hi Dale, >thank you for the valuable info, I am sorry with my terminology that I made >the mistake in explaining. > >1. As of now I can say that the port named in the SDP that I receive is the >port "to which" I send my RTP (this is no problem and works perfectly fine >when i use this as destination port to send my RTP). > >2. Just for double confirmation is it normal that the port "from which" I >send my RTP is irrelevant (then I really don't need to worry), because as >far as I can understand the port 5060 is already open and when I give this >port number (source port) from which I send my RTP works perfectly fine. > >if you say this is absolutely fine then I really don't need to worry. The >only thing I need to take care is that I should use my own defined RTP port >(which I have sent with my INVITE/SDP) to listen to incoming RTP packets. > >Correct me if I am wrong *Cheers* , Pranav Damele ** On 9 March 2011 10:33, <[email protected]>wrote: > Send Sip-implementors mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > or, via email, send a message with subject or body 'help' to > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Sip-implementors digest..." > > > Today's Topics: > > 1. Re: Audio Port problem (Siga) > 2. Re: Audio Port problem (Attila Sipos) > 3. Different SDP Session Version in 183 & 200 OK (Nitin Kapoor) > 4. Re: Different SDP Session Version in 183 & 200 OK > (Kevin P. Fleming) > 5. Re: Different SDP Session Version in 183 & 200 OK (Nitin Kapoor) > 6. Re: Different SDP Session Version in 183 & 200 OK - Email > found in subject (Johan DE CLERCQ) > 7. Re: [Sip] Different SDP Session Version in 183 & 200 OK > (Nitin Kapoor) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 8 Mar 2011 10:28:01 +0100 > From: Siga <[email protected]> > Subject: Re: [Sip-implementors] Audio Port problem > To: "Worley, Dale R (Dale)" <[email protected]> > Cc: "[email protected]" > <[email protected]> > Message-ID: > <[email protected]> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi Dale, > thank you for the valuable info, I am sorry with my terminology that I made > the mistake in explaining. > > 1. As of now I can say that the port named in the SDP that I receive is the > port "to which" I send my RTP (this is no problem and works perfectly fine > when i use this as destination port to send my RTP). > > 2. Just for double confirmation is it normal that the port "from which" I > send my RTP is irrelevant (then I really don't need to worry), because as > far as I can understand the port 5060 is already open and when I give this > port number (source port) from which I send my RTP works perfectly fine. > > if you say this is absolutely fine then I really don't need to worry. The > only thing I need to take care is that I should use my own defined RTP port > (which I have sent with my INVITE/SDP) to listen to incoming RTP packets. > > Correct me if I am wrong > > Regards > > > > On Mon, Mar 7, 2011 at 7:00 PM, Worley, Dale R (Dale) <[email protected] > >wrote: > > > ________________________________________ > > From: [email protected] [ > > [email protected]] On Behalf Of Siga [ > > [email protected]] > > > > I am parsing the audio port number which I get as SDP from my SIP Server. > I > > use this port number for sending my RTP Packets. > > _______________________________________________ > > > > You need to be careful with your terminology. The port named in the SDP > > that you receive is the port *to which* you must send your RTP. The port > > (on your system) *from which* you send RTP is irrelevant. Now, you may > > understand this, but what you wrote does not make that clear. Similarly, > > the port named in the SDP that you send is the port on your system on > which > > you will listen for RTP. > > > > Dale > > > > > ------------------------------ > > Message: 2 > Date: Tue, 8 Mar 2011 11:41:37 -0000 > From: "Attila Sipos" <[email protected]> > Subject: Re: [Sip-implementors] Audio Port problem > To: "Siga" <[email protected]>, "Worley, Dale R \(Dale\)" > <[email protected]> > Cc: [email protected] > Message-ID: > <[email protected]> > Content-Type: text/plain; charset="US-ASCII" > > >>2. Just for double confirmation is it normal that the port "from > which" I send my > >>RTP is irrelevant > > It is not normal. > It is not totally irrelevant. > For NAT traversal "symmetric RTP" is important. > See section 4 of tfc 4961: http://tools.ietf.org/rfc/rfc4961.txt > > Also some equipment may require that any received RTP has the same > source port > as the destination port for sent RTP. It's a sort of integrity check. > > > >>port 5060 is already open and when I give this port number (source > port) > >>from which I send my RTP works perfectly fine. > > It might work but it is not at all scaleable. > > Regards > > Attila > > > > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of > Siga > Sent: 08 March 2011 09:28 > To: Worley, Dale R (Dale) > Cc: [email protected] > Subject: Re: [Sip-implementors] Audio Port problem > > Hi Dale, > thank you for the valuable info, I am sorry with my terminology that I > made the mistake in explaining. > > 1. As of now I can say that the port named in the SDP that I receive is > the port "to which" I send my RTP (this is no problem and works > perfectly fine when i use this as destination port to send my RTP). > > 2. Just for double confirmation is it normal that the port "from which" > I send my RTP is irrelevant (then I really don't need to worry), because > as far as I can understand the port 5060 is already open and when I give > this port number (source port) from which I send my RTP works perfectly > fine. > > if you say this is absolutely fine then I really don't need to worry. > The only thing I need to take care is that I should use my own defined > RTP port (which I have sent with my INVITE/SDP) to listen to incoming > RTP packets. > > Correct me if I am wrong > > Regards > > > > On Mon, Mar 7, 2011 at 7:00 PM, Worley, Dale R (Dale) > <[email protected]>wrote: > > > ________________________________________ > > From: [email protected] [ > > [email protected]] On Behalf Of Siga [ > > [email protected]] > > > > I am parsing the audio port number which I get as SDP from my SIP > > Server. I use this port number for sending my RTP Packets. > > _______________________________________________ > > > > You need to be careful with your terminology. The port named in the > > SDP that you receive is the port *to which* you must send your RTP. > > The port (on your system) *from which* you send RTP is irrelevant. > > Now, you may understand this, but what you wrote does not make that > > clear. Similarly, the port named in the SDP that you send is the port > > > on your system on which you will listen for RTP. > > > > Dale > > > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > ------------------------------ > > Message: 3 > Date: Tue, 8 Mar 2011 16:48:28 -0500 > From: Nitin Kapoor <[email protected]> > Subject: [Sip-implementors] Different SDP Session Version in 183 & 200 > OK > To: [email protected] > Message-ID: > <[email protected]> > Content-Type: text/plain; charset=ISO-8859-1 > > Dear All, > > I have one call scenario where my termination is sending the SDP in 183 as > well as in 200 OK also. As far as i know if we are getting SDP in 183 > session progress then my UAC can ignore the SDP in 200 OK. Also most of the > time SDP is same. > > But here i noticed the slight difference of "Session Version". Here when my > termination is sending 188 Session Progress with SDP is sending the SDP as > below. > > I can see that my Termination is incrementing "*Session Version*" for SDP > in 183 & 200 OK in same dialog.. > > *183 with SDP* > > S_OWNER : o=TLPMSXP2 22660 *22660* IN IP4 69.90.230.210 > S_NAME : s=sip call > S_CONNECT : c=IN IP4 69.90.230.217 > TIME : t=0 0 > M_NAME : m=audio 59072 RTP/AVP 18 4 8 98 > > 200 OK with SDP: > > S_OWNER : o=TLPMSXP2 22660 *22661* IN IP4 69.90.230.210 > S_NAME : s=sip call > S_CONNECT : c=IN IP4 69.90.230.217 > TIME : t=0 0 > > Could anyone please let me know if that is okay to increment the session > version and if any supported document is there? > > Thanks, > Nitin > > > ------------------------------ > > Message: 4 > Date: Tue, 08 Mar 2011 15:54:51 -0600 > From: "Kevin P. Fleming" <[email protected]> > Subject: Re: [Sip-implementors] Different SDP Session Version in 183 & > 200 OK > To: [email protected] > Message-ID: <[email protected]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 03/08/2011 03:48 PM, Nitin Kapoor wrote: > > Dear All, > > > > I have one call scenario where my termination is sending the SDP in 183 > as > > well as in 200 OK also. As far as i know if we are getting SDP in 183 > > session progress then my UAC can ignore the SDP in 200 OK. Also most of > the > > time SDP is same. > > > > But here i noticed the slight difference of "Session Version". Here when > my > > termination is sending 188 Session Progress with SDP is sending the SDP > as > > below. > > > > I can see that my Termination is incrementing "*Session Version*" for > SDP > > in 183& 200 OK in same dialog.. > > > > *183 with SDP* > > > > S_OWNER : o=TLPMSXP2 22660 *22660* IN IP4 69.90.230.210 > > S_NAME : s=sip call > > S_CONNECT : c=IN IP4 69.90.230.217 > > TIME : t=0 0 > > M_NAME : m=audio 59072 RTP/AVP 18 4 8 98 > > > > 200 OK with SDP: > > > > S_OWNER : o=TLPMSXP2 22660 *22661* IN IP4 69.90.230.210 > > S_NAME : s=sip call > > S_CONNECT : c=IN IP4 69.90.230.217 > > TIME : t=0 0 > > > > Could anyone please let me know if that is okay to increment the session > > version and if any supported document is there? > > Was the to-tag in the 183 and 200 responses the same or had it changed? > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > > ------------------------------ > > Message: 5 > Date: Wed, 9 Mar 2011 02:24:42 -0500 > From: Nitin Kapoor <[email protected]> > Subject: Re: [Sip-implementors] Different SDP Session Version in 183 & > 200 OK > To: [email protected] > Cc: [email protected] > Message-ID: > <[email protected]> > Content-Type: text/plain; charset=ISO-8859-1 > > Hello All, > > Could any one please help me out on requested query as below. > > Thanks, > Nitin > > On Tue, Mar 8, 2011 at 4:48 PM, Nitin Kapoor <[email protected]> > wrote: > > > Dear All, > > > > I have one call scenario where my termination is sending the SDP in 183 > as > > well as in 200 OK also. As far as i know if we are getting SDP in 183 > > session progress then my UAC can ignore the SDP in 200 OK. Also most of > the > > time SDP is same. > > > > But here i noticed the slight difference of "Session Version". Here when > my > > termination is sending 188 Session Progress with SDP is sending the SDP > as > > below. > > > > I can see that my Termination is incrementing "*Session Version*" for > > SDP in 183 & 200 OK in same dialog.. > > > > *183 with SDP* > > > > S_OWNER : o=TLPMSXP2 22660 *22660* IN IP4 69.90.230.210 > > S_NAME : s=sip call > > S_CONNECT : c=IN IP4 69.90.230.217 > > TIME : t=0 0 > > M_NAME : m=audio 59072 RTP/AVP 18 4 8 98 > > > > 200 OK with SDP: > > > > S_OWNER : o=TLPMSXP2 22660 *22661* IN IP4 69.90.230.210 > > S_NAME : s=sip call > > S_CONNECT : c=IN IP4 69.90.230.217 > > TIME : t=0 0 > > > > Could anyone please let me know if that is okay to increment the session > > version and if any supported document is there? > > > > Thanks, > > Nitin > > > > > ------------------------------ > > Message: 6 > Date: Wed, 9 Mar 2011 08:35:08 +0100 > From: Johan DE CLERCQ <[email protected]> > Subject: Re: [Sip-implementors] Different SDP Session Version in 183 & > 200 OK - Email found in subject > To: Nitin Kapoor <[email protected]>, > "[email protected]" > <[email protected]> > Cc: "[email protected]" <[email protected]> > Message-ID: > <[email protected]> > Content-Type: text/plain; charset="us-ascii" > > I don't know if there's a document about this, but in my opinion you will > not encounter any problem with the incrementation. > > From: [email protected] [mailto: > [email protected]] On Behalf Of Nitin Kapoor > Sent: woensdag 9 maart 2011 8:25 > To: [email protected] > Cc: [email protected] > Subject: Re: [Sip-implementors] Different SDP Session Version in 183 & 200 > OK - Email found in subject > > > Hello All, > > Could any one please help me out on requested query as below. > > Thanks, > Nitin > > On Tue, Mar 8, 2011 at 4:48 PM, Nitin Kapoor <[email protected]> > wrote: > > > Dear All, > > > > I have one call scenario where my termination is sending the SDP in 183 > as > > well as in 200 OK also. As far as i know if we are getting SDP in 183 > > session progress then my UAC can ignore the SDP in 200 OK. Also most of > the > > time SDP is same. > > > > But here i noticed the slight difference of "Session Version". Here when > my > > termination is sending 188 Session Progress with SDP is sending the SDP > as > > below. > > > > I can see that my Termination is incrementing "*Session Version*" for > > SDP in 183 & 200 OK in same dialog.. > > > > *183 with SDP* > > > > S_OWNER : o=TLPMSXP2 22660 *22660* IN IP4 69.90.230.210 > > S_NAME : s=sip call > > S_CONNECT : c=IN IP4 69.90.230.217 > > TIME : t=0 0 > > M_NAME : m=audio 59072 RTP/AVP 18 4 8 98 > > > > 200 OK with SDP: > > > > S_OWNER : o=TLPMSXP2 22660 *22661* IN IP4 69.90.230.210 > > S_NAME : s=sip call > > S_CONNECT : c=IN IP4 69.90.230.217 > > TIME : t=0 0 > > > > Could anyone please let me know if that is okay to increment the session > > version and if any supported document is there? > > > > Thanks, > > Nitin > > > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > ________________________________ > > Geen virus gevonden in dit bericht. > Gecontroleerd door AVG - www.avg.com<http://www.avg.com> > Versie: 10.0.1204 / Virusdatabase: 1497/3492 - datum van uitgifte: 03/08/11 > > > ------------------------------ > > Message: 7 > Date: Wed, 9 Mar 2011 04:33:27 -0500 > From: Nitin Kapoor <[email protected]> > Subject: Re: [Sip-implementors] [Sip] Different SDP Session Version in > 183 & 200 OK > To: Ashish Saxena <[email protected]> > Cc: [email protected], [email protected] > Message-ID: > <[email protected]> > Content-Type: text/plain; charset=ISO-8859-1 > > Hello Ashish, > > Here is the mline for both the messages. > > 183: > > Media Description, name and address (m): audio 43888 RTP/AVP 18 > > 200 OK: > > Media Description, name and address (m): audio 43888 RTP/AVP 18 > > Thanks, > Nitin Kapoor > > > On Wed, Mar 9, 2011 at 3:25 AM, Ashish Saxena <[email protected] > >wrote: > > > what is the mline of 200OK SDP. > > > > Regards > > Ashish Saxena > > (www.aricent.com) > > ________________________________________ > > From: [email protected] [[email protected]] On Behalf Of Nitin > > Kapoor [[email protected]] > > Sent: Wednesday, March 09, 2011 12:54 PM > > To: [email protected] > > Cc: [email protected] > > Subject: Re: [Sip] Different SDP Session Version in 183 & 200 OK > > > > Hello All, > > > > Could any one please help me out on requested query as below. > > > > Thanks, > > Nitin > > > > On Tue, Mar 8, 2011 at 4:48 PM, Nitin Kapoor <[email protected] > > <mailto:[email protected]>> wrote: > > Dear All, > > > > I have one call scenario where my termination is sending the SDP in 183 > as > > well as in 200 OK also. As far as i know if we are getting SDP in 183 > > session progress then my UAC can ignore the SDP in 200 OK. Also most of > the > > time SDP is same. > > > > But here i noticed the slight difference of "Session Version". Here when > my > > termination is sending 188 Session Progress with SDP is sending the SDP > as > > below. > > > > I can see that my Termination is incrementing "Session Version" for SDP > > in 183 & 200 OK in same dialog.. > > > > 183 with SDP > > > > S_OWNER : o=TLPMSXP2 22660 22660 IN IP4 69.90.230.210 > > S_NAME : s=sip call > > S_CONNECT : c=IN IP4 69.90.230.217 > > TIME : t=0 0 > > M_NAME : m=audio 59072 RTP/AVP 18 4 8 98 > > > > 200 OK with SDP: > > > > S_OWNER : o=TLPMSXP2 22660 22661 IN IP4 69.90.230.210 > > S_NAME : s=sip call > > S_CONNECT : c=IN IP4 69.90.230.217 > > TIME : t=0 0 > > > > Could anyone please let me know if that is okay to increment the session > > version and if any supported document is there? > > > > Thanks, > > Nitin > > > > > > "DISCLAIMER: This message is proprietary to Aricent and is intended > solely > > for the use of the individual to whom it is addressed. It may contain > > privileged or confidential information and should not be circulated or > used > > for any purpose other than for what it is intended. If you have received > > this message in error, please notify the originator immediately. If you > are > > not the intended recipient, you are notified that you are strictly > > prohibited from using, copying, altering, or disclosing the contents of > this > > message. Aricent accepts no responsibility for loss or damage arising > from > > the use of the information transmitted by this email including damage > from > > virus." > > > > > ------------------------------ > > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > End of Sip-implementors Digest, Vol 96, Issue 9 > *********************************************** > _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
