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DocLarge
Premium
join:2004-09-08
kudos:1

SIP Audio Ports

Just needed to ask a quick question outside the scope of another thread I have running for assistance....

Which ports (other than 5000 - 5084) are responsible for "sending" audio in FreePBX? For example it's know the following config file is responsible for receiving audio:

rtp.conf (Receive Audio)

rtpstart= 10000
rtpend= 20000

Is there an equivalent config file for "sending" audio or are ports 5000 - 5084 the "send audio" ports?

Jay


Stewart

join:2005-07-13
kudos:25

Every UDP packet has both a "source" port and a "destination" port. Packets sent from your system have the source port determined by local software, e.g. the rtp.conf file; the destination port is determined by the remote system. The SDP sent from the remote system tells your system which destination port to use.

For RTP that you receive, the source port is determined by the remote system. The destination port, selected by your system in the range specified in rtp.conf, is transmitted in the SDP so the remote system knows what port to use.

A convention for RTP is to use matching port numbers -- the source port for RTP you send is the same as the destination port for RTP received on that call. Likewise, the remote system will source RTP it sends from the same port as used for receiving RTP from you.


DocLarge
Premium
join:2004-09-08
kudos:1

Forgive me Stew, that one went by me *whoosh*

Is there a graphical illustration of what you just said?

Jay


Stewart

join:2005-07-13
kudos:25

Sorry, but if your PBX were on an unrestricted public IP, you wouldn't care about port numbers -- it would just work (though it might not be secure). So, I assume that you need to configure or are having trouble with a router or firewall. To do that intelligently, you need to have an understanding of basic networking concepts. A picture won't do. Lots of pictures might, but I don't offhand know of such a book or site to recommend.

Surely, you understand the concept of IP addresses. Each IP packet has both a source and a destination address. The destination is used by routers to get the packet to the right place. The source tells the recipient where it came from. You can think of ports as sub-addresses, like an apartment or suite number on a physical address. Port numbers are used by NAT routers to get a packet to the proper host on your LAN, by the operating system on that host to get it to the proper application (e.g. Asterisk), and by the application itself, e.g. so Asterisk knows to which call an RTP packet belongs.

SIP uses SDP (Session Description Protocol) to tell the other end where (IP address and port) to send RTP for that call. It limits its choices to the range specified in rtp.conf. In general, you have no control over the port number used by the remote system, so your system must be set up to send RTP to any requested port number, and to receive RTP from that port number as well.

If you are still confused, get a good book or visit some educational Web sites. Sorry, there is not yet a Manga Guide to VoIP. There is, though, a Manga Guide to Databases, so it may be coming



Trev
IP Telephony Addict
Premium
join:2009-06-29
Victoria, BC
kudos:6
reply to DocLarge

I wonder if you're asking the right question. I'm not sure what this other thread you refer to is, but if you could let us know the problem you're trying to solve, perhaps we could better understand and help you formulate the right question and provide a more useful answer.

Steward is definitely correct on a lot of points, however I get a strong impression that you're not asking the right question here.
--
Wondering what I do? Find out at »www.digitalcon.ca


DocLarge
Premium
join:2004-09-08
kudos:1

1 edit
reply to DocLarge

*Heh* It's kewl, Stew... I'm not a "voice" engineer, but I've learned more voice principles in the last two months to tide me over for a bit. I just re-read what you posted and now that I've had a nap, it translated better (been at this for 17hrs and counting).

Trev, I did some background searching and found the answer I was looking for regarding the voice ports.

The long and short (that I've just discovered) is that the "new" box I've been configuring for use with FreePBX doesn't "go along for the ride" as easily as my outdated Compaq Presario did. I've just found that settings that "automagically" forwarded with the Compaq didn't with the Zotac, so there went my entire day Oh well, now I know.

In the meantime, I've just finished configuring a virtual freepbx server on my Vaio laptop and I'm going to see if the same anomolies occur (I'll let you guys know).

Thanks for assist...

Jay

CERTIFIED: CCNA, Security+, Network+, Server+, I.T. Project+

(As I said, I'm not a "voice" engineer, but I'm well on my way) Hell, after being exposed to FreePBX, I'm motivated to check out the Cisco voice track *heh*


DocLarge
Premium
join:2004-09-08
kudos:1
reply to DocLarge

Boom, baby!!! The virtual pbx I configured on my laptop works without any additional configuration... I just called my cellphone and the call went over without any dropouts! Audio was bi-directional and not one way; sweet!!

Unfortunately this means I'll have to dig a little deeper in order to find out why my $300 Zotac box is being difficult in running Freepbx.


Dan_voip

join:2007-01-03
Saint-Hubert, QC
kudos:4
Reviews:
·TekSavvy DSL

From what I know 5000 - 5084 are used for registration/signaling and 10000-20000 for voice in and out. I'm guessing you want to do port forwarding and that's why you need the port numbers.
With your background you'll be more then happy installing a plain vanilla Asterisk where you have full control of what you're doing and don't have to guess what the GUI is doing in background. For me it's like using a linux/unix console instead of a GUI:)



nunya
Premium,MVM
join:2000-12-23
O Fallon, MO
kudos:12
reply to DocLarge

There have been certain unexplainable issues reported with certain versions of Asterisk and Intel Atom based pc's.

Are you still using PBXIAF?
--
Looks like Reverend Wright got his wish - God Damn America.


DocLarge
Premium
join:2004-09-08
kudos:1

1 edit

Dan,

It's to the point where I'm just accepting the fact I need to go ahead and learn Linux as opposed to pulling "quick fix IOS inputs" off the web to keep things moving

Nunya, I'm still using PIAF and I can honestly say with the exception of this snafu, I love the Hell out of it ! I'm going to strip this thing down again and try the 1.6 load to see if the problem still persists. On another note, I'm getting "killer" performance out of my virtual freepbx server (I'm been calling around on it all day).

I bridged the intel wireless card on the notebook, assigned an ip address to the virtual, and "voila!" Call quality is excellent from what everyone is saying and I'm noticing it as well. Hmmmmm... If I can't get Freepbx to load natively on the Zotac, I may just load an OS and then put a virtual load of freepbx on it as well (now that's not a bad idea...)

More to follow...

Jay