Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etc....I noticed that each call uses four UDP port for RTP. Here is an example of port for a call I made: 10200 10201 10504 10505 Seems like they are random in pair. I have a restriction of 10000-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110114/9c8a960c/attachment.htm>
I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B <bruceb444 at gmail.com> wrote:> Hi Everyone, > > I am just tweaking a pfSense router and learning lots about NAT etc....I > noticed that each call uses four UDP port for RTP. Here is an example of > port for a call I made: > > 10200 > 10201 > 10504 > 10505 > > Seems like they are random in pair. I have a restriction of 10000-11000 in > my rtp.conf so that makes sense. But why use 4 ports per call? is that part > of SIP RFC? > > Thanks >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110114/09e0253f/attachment.htm>
RTP always uses a random even numbered port, then RTCP will use the next port, which will always be odd numbered. Symmetric RTP only needs two ports, while asymmetric RTP uses four. http://www.armware.dk/RFC/rfc/rfc4961.html On Fri, Jan 14, 2011 at 12:44 PM, Bruce B <bruceb444 at gmail.com> wrote:> I mean part of RTP RFC? > > > On Fri, Jan 14, 2011 at 2:41 PM, Bruce B <bruceb444 at gmail.com> wrote: > >> Hi Everyone, >> >> I am just tweaking a pfSense router and learning lots about NAT etc....I >> noticed that each call uses four UDP port for RTP. Here is an example of >> port for a call I made: >> >> 10200 >> 10201 >> 10504 >> 10505 >> >> Seems like they are random in pair. I have a restriction of 10000-11000 in >> my rtp.conf so that makes sense. But why use 4 ports per call? is that part >> of SIP RFC? >> >> Thanks >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110114/07ff826c/attachment.htm>
Thanks guys. I am not sure whether that call was asymmetric or not but I saw 4 ports open. It could be that the other two ports were remnant of another channel even though I doubt it. Now, when I tried again, it is only 2 ports that is opened like you mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use the symmetric method or is the asymmetric method used as well by some media servers? The reason why I am asking is because there are many many online responses that there is 4 ports needed per call and make sure you keep enough ports open, blah blah... Thanks again On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen <solstars1 at gmail.com> wrote:> RTP always uses a random even numbered port, then RTCP will use the next > port, which will always be odd numbered. Symmetric RTP only needs two > ports, while asymmetric RTP uses four. > > http://www.armware.dk/RFC/rfc/rfc4961.html > > > > On Fri, Jan 14, 2011 at 12:44 PM, Bruce B <bruceb444 at gmail.com> wrote: > >> I mean part of RTP RFC? >> >> >> On Fri, Jan 14, 2011 at 2:41 PM, Bruce B <bruceb444 at gmail.com> wrote: >> >>> Hi Everyone, >>> >>> I am just tweaking a pfSense router and learning lots about NAT etc....I >>> noticed that each call uses four UDP port for RTP. Here is an example of >>> port for a call I made: >>> >>> 10200 >>> 10201 >>> 10504 >>> 10505 >>> >>> Seems like they are random in pair. I have a restriction of 10000-11000 >>> in my rtp.conf so that makes sense. But why use 4 ports per call? is that >>> part of SIP RFC? >>> >>> Thanks >>> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110114/93abef83/attachment.htm>
Danny Nicholas
2011-Jan-14 20:33 UTC
[asterisk-users] Why are 4 ports used for a single call?
Hurray for Microsoft Outlook (for creating this whole top-post thread).
Just my .02; The other two ports must have been a remnant of another
channel; as for the 4 ports - I think that the 4 port requirement is
probably for "niceties" like conferencing and transfers.
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bruce B
Sent: Friday, January 14, 2011 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Why are 4 ports used for a single call?
Thanks guys. I am not sure whether that call was asymmetric or not but I saw
4 ports open. It could be that the other two ports were remnant of another
channel even though I doubt it.
Now, when I tried again, it is only 2 ports that is opened like you
mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use the
symmetric method or is the asymmetric method used as well by some media
servers?
The reason why I am asking is because there are many many online responses
that there is 4 ports needed per call and make sure you keep enough ports
open, blah blah...
Thanks again
On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen <solstars1 at gmail.com>
wrote:
RTP always uses a random even numbered port, then RTCP will use the next
port, which will always be odd numbered. Symmetric RTP only needs two
ports, while asymmetric RTP uses four.
http://www.armware.dk/RFC/rfc/rfc4961.html
On Fri, Jan 14, 2011 at 12:44 PM, Bruce B <bruceb444 at gmail.com> wrote:
I mean part of RTP RFC?
On Fri, Jan 14, 2011 at 2:41 PM, Bruce B <bruceb444 at gmail.com> wrote:
Hi Everyone,
I am just tweaking a pfSense router and learning lots about NAT etc....I
noticed that each call uses four UDP port for RTP. Here is an example of
port for a call I made:
10200
10201
10504
10505
Seems like they are random in pair. I have a restriction of 10000-11000 in
my rtp.conf so that makes sense. But why use 4 ports per call? is that part
of SIP RFC?
Thanks
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Got it. Thanks. Makes sense to keep an extra two in mind for conference etc.... Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. On Fri, Jan 14, 2011 at 3:33 PM, Danny Nicholas <danny at debsinc.com> wrote:> Hurray for Microsoft Outlook (for creating this whole top-post thread). > Just my .02; The other two ports must have been a remnant of another > channel; as for the 4 ports ? I think that the 4 port requirement is > probably for ?niceties? like conferencing and transfers. > > > ------------------------------ > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *Bruce B > *Sent:* Friday, January 14, 2011 2:15 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Why are 4 ports used for a single call? > > > > Thanks guys. I am not sure whether that call was asymmetric or not but I > saw 4 ports open. It could be that the other two ports were remnant of > another channel even though I doubt it. > > > > Now, when I tried again, it is only 2 ports that is opened like you > mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use > the symmetric method or is the asymmetric method used as well by some media > servers? > > > > The reason why I am asking is because there are many many > online responses that there is 4 ports needed per call and make sure you > keep enough ports open, blah blah... > > > > Thanks again > > On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen <solstars1 at gmail.com> wrote: > > RTP always uses a random even numbered port, then RTCP will use the next > port, which will always be odd numbered. Symmetric RTP only needs two > ports, while asymmetric RTP uses four. > > http://www.armware.dk/RFC/rfc/rfc4961.html > > > On Fri, Jan 14, 2011 at 12:44 PM, Bruce B <bruceb444 at gmail.com> wrote: > > I mean part of RTP RFC? > > > > On Fri, Jan 14, 2011 at 2:41 PM, Bruce B <bruceb444 at gmail.com> wrote: > > Hi Everyone, > > > > I am just tweaking a pfSense router and learning lots about NAT etc....I > noticed that each call uses four UDP port for RTP. Here is an example of > port for a call I made: > > > > 10200 > > 10201 > > 10504 > > 10505 > > > > Seems like they are random in pair. I have a restriction of 10000-11000 in > my rtp.conf so that makes sense. But why use 4 ports per call? is that part > of SIP RFC? > > > > Thanks > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110114/f4275bc7/attachment.htm>
Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060 right? and why are there recommendations of opening 5000-5082 UDP for SIP along with 5060 TCP? Are there any "niceties" to that as well? maybe video transmission stuff? Thanks again, On Fri, Jan 14, 2011 at 4:12 PM, Bruce B <bruceb444 at gmail.com> wrote:> Got it. Thanks. Makes sense to keep an extra two in mind for conference > etc.... > > Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. > > > On Fri, Jan 14, 2011 at 3:33 PM, Danny Nicholas <danny at debsinc.com> wrote: > >> Hurray for Microsoft Outlook (for creating this whole top-post thread). >> Just my .02; The other two ports must have been a remnant of another >> channel; as for the 4 ports ? I think that the 4 port requirement is >> probably for ?niceties? like conferencing and transfers. >> >> >> ------------------------------ >> >> *From:* asterisk-users-bounces at lists.digium.com [mailto: >> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Bruce B >> *Sent:* Friday, January 14, 2011 2:15 PM >> >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* Re: [asterisk-users] Why are 4 ports used for a single call? >> >> >> >> Thanks guys. I am not sure whether that call was asymmetric or not but I >> saw 4 ports open. It could be that the other two ports were remnant of >> another channel even though I doubt it. >> >> >> >> Now, when I tried again, it is only 2 ports that is opened like you >> mentioned, even RTP port, and RTP port +1. So, does Asterisk usually use >> the symmetric method or is the asymmetric method used as well by some media >> servers? >> >> >> >> The reason why I am asking is because there are many many >> online responses that there is 4 ports needed per call and make sure you >> keep enough ports open, blah blah... >> >> >> >> Thanks again >> >> On Fri, Jan 14, 2011 at 2:57 PM, Gary Allen <solstars1 at gmail.com> wrote: >> >> RTP always uses a random even numbered port, then RTCP will use the next >> port, which will always be odd numbered. Symmetric RTP only needs two >> ports, while asymmetric RTP uses four. >> >> http://www.armware.dk/RFC/rfc/rfc4961.html >> >> >> On Fri, Jan 14, 2011 at 12:44 PM, Bruce B <bruceb444 at gmail.com> wrote: >> >> I mean part of RTP RFC? >> >> >> >> On Fri, Jan 14, 2011 at 2:41 PM, Bruce B <bruceb444 at gmail.com> wrote: >> >> Hi Everyone, >> >> >> >> I am just tweaking a pfSense router and learning lots about NAT etc....I >> noticed that each call uses four UDP port for RTP. Here is an example of >> port for a call I made: >> >> >> >> 10200 >> >> 10201 >> >> 10504 >> >> 10505 >> >> >> >> Seems like they are random in pair. I have a restriction of 10000-11000 in >> my rtp.conf so that makes sense. But why use 4 ports per call? is that part >> of SIP RFC? >> >> >> >> Thanks >> >> >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110114/fb19afad/attachment-0001.htm>