Re: [asterisk-users] Why are 4 ports used for a single call?
I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI 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 1-11000 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
Re: [asterisk-users] Why are 4 ports used for a single call?
Each media stream will use two, one for RTP and one for RTCP. In your case 10200/10201 are an RTP/RTCP pair, and 10504/10505 are another pair. RTP is always and even numbered port, and RTCP is always RTP port + 1. Yes, it's in the RFC for RTP. The fact that you have two pairs means that two media streams are being negotiated, perhaps one for audio and one for video? Your phone config or wireshark captures will tell you for certain. Of course, I'm assuming that those ports are for one endpoint (phone). If one pair is for caller and one pair is for callee, then this is a normal simplest scenario, one pair for each side. You didn't specify whose ports they were. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Friday, January 14, 2011 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why are 4 ports used for a single call? I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI 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 1-11000 in my rtp.conf so that makes sense. But why use 4 ports per call? is that part of SIP RFC? Thanks smime.p7s Description: S/MIME cryptographic signature -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
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 bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI 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 1-11000 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
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 solsta...@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 bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI 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 1-11000 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
Re: [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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 solsta...@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 bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI 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 1-11000 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
Re: [asterisk-users] Why are 4 ports used for a single call?
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 bruceb...@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 da...@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-boun...@lists.digium.com [mailto: asterisk-users-boun...@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 solsta...@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 bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI 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 1-11000 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 -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
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 da...@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-boun...@lists.digium.com [mailto: asterisk-users-boun...@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 solsta...@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 bruceb...@gmail.com wrote: I mean part of RTP RFC? On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote: Hi Everyone, I am just tweaking a pfSense router and learning lots about NAT etcI 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 1-11000 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 -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
On Friday 14 January 2011 15:12:29 Bruce B wrote: Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. http://www.justfuckinggoogleit.com/search.pl?query=top+posting It's why most of the experts in here ignore your posts. If you haven't got the good sense to follow etiquette, the Delete key becomes the first line of defense. -- Tilghman -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
On 01/14/2011 4:19 PM, Bruce B wrote: 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? More likely, it's because only one client behind NAT can use port 5060, so other clients need to use other ports. Could be another reason, though. Tom -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
So, simply pressing Reply and typing in the first line (using gmail webmail without any clients) is a sin here? How is that top posting??? probably your clients reading that way? On Fri, Jan 14, 2011 at 5:13 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Friday 14 January 2011 15:12:29 Bruce B wrote: Off topic - what is top post? I am using gmail + chrome - no ugly Outlook. http://www.justfuckinggoogleit.com/search.pl?query=top+posting It's why most of the experts in here ignore your posts. If you haven't got the good sense to follow etiquette, the Delete key becomes the first line of defense. -- Tilghman -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
On Fri, 14 Jan 2011 17:29:26 -0500, Tom Rymes try...@rymes.com wrote: On 01/14/2011 4:19 PM, Bruce B wrote: 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? More likely, it's because only one client behind NAT can use port 5060, so other clients need to use other ports. Could be another reason, though. FWIW, by default, GrandStream Budget phones use UDP 5004 for RTP. -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
On Jan 14, 2011, at 5:24 PM, Bruce B wrote: So, simply pressing Reply and typing in the first line (using gmail webmail without any clients) is a sin here? How is that top posting??? probably your clients reading that way? It may be a sin here, but it is certainly impolite many places, and illogical everywhere. This is because we normally read top to bottom, but top-posting forces you to read bottom to top. Tom -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
You really want to read the LONG LONG signature from some people before you read the actual latest message? I don't know about thatI guess it's a preference. Back to my other questions, now that UDP is clear for me, what ports does SIP require? TCP/UDP 5060 ? 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 On Fri, Jan 14, 2011 at 6:32 PM, Tom Rymes try...@rymes.com wrote: On Jan 14, 2011, at 5:24 PM, Bruce B wrote: So, simply pressing Reply and typing in the first line (using gmail webmail without any clients) is a sin here? How is that top posting??? probably your clients reading that way? It may be a sin here, but it is certainly impolite many places, and illogical everywhere. This is because we normally read top to bottom, but top-posting forces you to read bottom to top. Tom -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
On Jan 14, 2011, at 6:45 PM, Bruce B wrote: You really want to read the LONG LONG signature from some people before you read the actual latest message? I don't know about thatI guess it's a preference. Suffice it to say, Bruce, this subject has been hashed over thousands, nay, hundreds of thousands of times, and I doubt anything new can be had from doing it again. FYI, It is also considered good etiquette to remove any non-relevant information from the quoted text to keep it short and easy to parse, especially removing the automatically generated footers from the list. As for your question about ports (see, I can stay on topic occasionally!), someone already mentioned something about some equipment using 5004 for RTP, IIRC, and I mentioned the common use of 5061, 5062, 5063, etc for multiple SIP clients behind NAT. There may be other reasons, too. Tom -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
On Fri, Jan 14, 2011 at 6:53 PM, Tom Rymes try...@rymes.com wrote: On Jan 14, 2011, at 6:45 PM, Bruce B wrote: You really want to read the LONG LONG signature from some people before you read the actual latest message? I don't know about thatI guess it's a preference. Suffice it to say, Bruce, this subject has been hashed over thousands, nay, hundreds of thousands of times, and I doubt anything new can be had from doing it again. FYI, It is also considered good etiquette to remove any non-relevant information from the quoted text to keep it short and easy to parse, especially removing the automatically generated footers from the list. As for your question about ports (see, I can stay on topic occasionally!), someone already mentioned something about some equipment using 5004 for RTP, IIRC, and I mentioned the common use of 5061, 5062, 5063, etc for multiple SIP clients behind NAT. There may be other reasons, too. Tom Thanks. That is in both TCP and UDP for SIP right? or simply UDP would do it as well? I am talking strictly in case of Asterisk. -Bruce -- _ -- 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
Re: [asterisk-users] Why are 4 ports used for a single call?
On Jan 14, 2011, at 7:12 PM, Bruce B wrote: Thanks. That is in both TCP and UDP for SIP right? or simply UDP would do it as well? I am talking strictly in case of Asterisk. Asterisk 1.6 and newer support SIP over TCP. Older versions were UDP only, IIRC. Tom -- _ -- 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
