21 Sep
2005
21 Sep
'05
6:19 p.m.
First off, here is my voice network layout currently:
http://webdev.digitalpath.net/~rayvd/voice/voice_network2.png
We're using Asterisk for voicemail, call routing (for long-distance, LNP, etc)
and SER/rtpproxy at the other end which handles NAT onto private networks
where customer's exist.
This setup works fairly well for the most part, except that Asterisk does not
have a jitter buffer. I would like to make use of rtpproxy (or mediaproxy)
for their jitter buffer on both ends of our voice links here. To me, that
would mean shoving a SER/rtpproxy combo between Asterisk and our provider
network. Possibly on the same server.
I could easily throw up a SER installation, but I'm trying to figure out if
there's any way to leave Asterisk in the SIP path but remove it from the media
path (have RTP just go straight from SER/rtpproxy to my provider's RTP proxy).
Have any of you set up a scenario somewhat like this? Any recommendations?
Asterisk CVS-HEAD does let you apply a patch and get some very alpha jitter
buffer support. But CVS HEAD doesn't work reliably for me at all currently so
I'm sticking with the latest stable release of 1.0.x series.
I'm running SER 0.9.3 FYI on the SER proxies I have set up currently.
Thanks.
--
Ray Van Dolson
Linux/Unix Systems Administrator
Digital Path, Inc.
23 Sep
23 Sep
5:17 a.m.
Ray,
Jitter buffers in VoIP are only of use at the point where the packets are
converted into the final PCM bit stream - a TDM or DAC interface in most
cases.
The problem with your proposed solution is that the output of mediaproxy,
even if it is de-jittered, will be subject to jitter once again on the way
to the endpoint where the packet connection is converted to voice.
Mark
On 9/21/05, Ray Van Dolson <rayvd(a)digitalpath.net> wrote:
First off, here is my voice network layout currently:
http://webdev.digitalpath.net/~rayvd/voice/voice_network2.png
We're using Asterisk for voicemail, call routing (for long-distance, LNP,
etc)
and SER/rtpproxy at the other end which handles NAT onto private networks
where customer's exist.
This setup works fairly well for the most part, except that Asterisk does
not
have a jitter buffer. I would like to make use of rtpproxy (or mediaproxy)
for their jitter buffer on both ends of our voice links here. To me, that
would mean shoving a SER/rtpproxy combo between Asterisk and our provider
network. Possibly on the same server.
I could easily throw up a SER installation, but I'm trying to figure out
if
there's any way to leave Asterisk in the SIP path but remove it from the
media
path (have RTP just go straight from SER/rtpproxy to my provider's RTP
proxy).
Have any of you set up a scenario somewhat like this? Any recommendations?
Asterisk CVS-HEAD does let you apply a patch and get some very alpha
jitter
buffer support. But CVS HEAD doesn't work reliably for me at all currently
so
I'm sticking with the latest stable release of 1.0.x series.
I'm running SER 0.9.3 FYI on the SER proxies I have set up currently.
Thanks.
--
Ray Van Dolson
Linux/Unix Systems Administrator
Digital Path, Inc.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
5:26 a.m.
Ray,
Just looking at your diagram again it seems puzzling as to why the lack of
a jitter buffer in Asterisk would be a problem for you at all. You are not
using Asterisk as a gateway.
Mark
On 9/21/05, Ray Van Dolson <rayvd(a)digitalpath.net> wrote:
First off, here is my voice network layout currently:
http://webdev.digitalpath.net/~rayvd/voice/voice_network2.png
We're using Asterisk for voicemail, call routing (for long-distance, LNP,
etc)
and SER/rtpproxy at the other end which handles NAT onto private networks
where customer's exist.
This setup works fairly well for the most part, except that Asterisk does
not
have a jitter buffer. I would like to make use of rtpproxy (or mediaproxy)
for their jitter buffer on both ends of our voice links here. To me, that
would mean shoving a SER/rtpproxy combo between Asterisk and our provider
network. Possibly on the same server.
I could easily throw up a SER installation, but I'm trying to figure out
if
there's any way to leave Asterisk in the SIP path but remove it from the
media
path (have RTP just go straight from SER/rtpproxy to my provider's RTP
proxy).
Have any of you set up a scenario somewhat like this? Any recommendations?
Asterisk CVS-HEAD does let you apply a patch and get some very alpha
jitter
buffer support. But CVS HEAD doesn't work reliably for me at all currently
so
I'm sticking with the latest stable release of 1.0.x series.
I'm running SER 0.9.3 FYI on the SER proxies I have set up currently.
Thanks.
--
Ray Van Dolson
Linux/Unix Systems Administrator
Digital Path, Inc.
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
26 Oct
26 Oct
11:21 p.m.
Mark, somehow I missed the replies to my original email and just stumbled
across them on the list archives.
The problem in the situation below is that there is jitter between SER and
Asterisk. Since Asterisk doesn't compensate for this, the person on the other
end of a call with one of our PAP2-NA customers hears choppiness. The jitter
buffer on the PAP2-NA helps deal with any jitter in the other direction.
Just trying to figure out if indeed rtpproxy or mediaproxy have a jitter
buffer built in. I could potentially throw another SER install nearer to the
Asterisk box to deal with the jitter and then hand off to Asterisk.
I'd love to get Asterisk completely out of the media path and just use it for
centralized call routing, voicemail, etc, but I'm not sure how to tell
Asterisk to tell other SIP clients to use my SER boxes for the media path
instead. In any case, that's a question for the Asterisk list. :-)
Basically.. anyone know if rtpproxy and/or mediaproxy have jitter buffers? :)
Ray
On Wed, Sep 21, 2005 at 09:19:53AM -0700, Ray Van Dolson wrote:
First off, here is my voice network layout currently:
http://webdev.digitalpath.net/~rayvd/voice/voice_network2.png
We're using Asterisk for voicemail, call routing (for long-distance, LNP, etc)
and SER/rtpproxy at the other end which handles NAT onto private networks
where customer's exist.
This setup works fairly well for the most part, except that Asterisk does not
have a jitter buffer. I would like to make use of rtpproxy (or mediaproxy)
for their jitter buffer on both ends of our voice links here. To me, that
would mean shoving a SER/rtpproxy combo between Asterisk and our provider
network. Possibly on the same server.
I could easily throw up a SER installation, but I'm trying to figure out if
there's any way to leave Asterisk in the SIP path but remove it from the media
path (have RTP just go straight from SER/rtpproxy to my provider's RTP proxy).
Have any of you set up a scenario somewhat like this? Any recommendations?
Asterisk CVS-HEAD does let you apply a patch and get some very alpha jitter
buffer support. But CVS HEAD doesn't work reliably for me at all currently so
I'm sticking with the latest stable release of 1.0.x series.
I'm running SER 0.9.3 FYI on the SER proxies I have set up currently.
Thanks.
27 Oct
27 Oct
1:26 a.m.
Ray Van Dolson wrote:
Basically.. anyone know if rtpproxy and/or mediaproxy have jitter buffers? :)
No jitter buffers in rtpproxy/mediaproxy. But if you are using ZAP
channels on Asterisk you can set jitter buffers there.
--
Andres
Network Admin
http://www.telesip.net
28 Oct
28 Oct
12:54 a.m.
On Wed, Oct 26, 2005 at 02:21:02PM -0700, Ray Van Dolson wrote:
I'd love to get Asterisk completely out of the
media path and just use it for
centralized call routing, voicemail, etc, but I'm not sure how to tell
Asterisk to tell other SIP clients to use my SER boxes for the media path
instead. In any case, that's a question for the Asterisk list. :-)
Finally figured out a way to get Asterisk completely out of the media path
(well, once the call has been established anyways).
SER rocks.
Ray
12:58 a.m.
Hi Ray,
Can you tell the rest of us how you did it?
Ta,
Pat
On Thu, 2005-10-27 at 15:54 -0700, Ray Van Dolson wrote:
On Wed, Oct 26, 2005 at 02:21:02PM -0700, Ray Van
Dolson wrote:
I'd love to get Asterisk completely out of
the media path and just use it for
centralized call routing, voicemail, etc, but I'm not sure how to tell
Asterisk to tell other SIP clients to use my SER boxes for the media path
instead. In any case, that's a question for the Asterisk list. :-)
Finally figured out a way to get Asterisk completely out of the media path
(well, once the call has been established anyways).
SER rocks.
Ray
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
1:57 a.m.
http://webdev.digitalpath.net/~rayvd/voice/ser_asterisk.cfg
Is my Asterisk configuration file.
Basically:
[SIP ATA] <--- 172.25.163.0/24 IP
[Customer CPE] <--- 10.0.0.0/8 IP
[SER] <--- 10.0.0.0/8 IP and Public Internet IP
[Asterisk] <--- Public Internet IP
Asterisk and SER are usually far from each other.
In my setup, Asterisk normally talks to my ISP's SIP/PSTN gateway and then
passes RTP on to my ISP's media gateway. So the normal flow of RTP is:
SIP ATA -> SER -> Asterisk -> Media Gateway -> PSTN
Some of our internet connections between SER and Asterisk can be a bit
finnicky at times and introduce jitter. Since Asterisk doesn't have a
(working) jitter buffer currently, any benefits from the jitter buffer on my
ISP's Media Gateway cannot really be realized.
So I set up the config as in my cfg above and set canreinvite to yes for the
ATA in my sip.conf on Asterisk (also make sure your Dial() in Asterisk's
extensions.conf doesn't have a t or T in the flags list). Then the following
happens:
1. I make a call from my SIP ATA to a PSTN number.
2. INVITE to SER
3. Normal negotiation happens. RTP is flowing as in my original flow above
(with Asterisk in the media path)
4. Asterisk now sends an INVITE to SER with an SDP payload specifying the IP
of my ISP's media gateway. This is a public out on the Internet.
5. This requests hits the loose route portion of my route {} block.
6. I rewrite the media IP to the 10.0.0.0/8 IP on my SER server, but also
specify the "R" flag to "trust" the IP originially passed by
Asterisk.
This allows SER to shift it's RTP stream to the Media Gateway's IP
address.
7. I use t_on_reply(2) because I want the reply from my SIP ATA for this
re-INVITE to hit my onreply_route[2] function.
8. The re-INVITE goes out to the SIP ATA and back comes the 200 OK response
with an SDP payload. This reply hits onreply_route[2] and I fix the media
source IP here to have the IP of the SER server.
At this point, Asterisk is now out of the media path completely!
Hope that was somewhat helpful. Don't frown too much on my messy ser.cfg
file. :-)
Ray
On Fri, Oct 28, 2005 at 11:58:22AM +1300, Patrick Jordan-Smith wrote:
Hi Ray,
Can you tell the rest of us how you did it?
Ta,
Pat
30 Oct
30 Oct
3:55 p.m.
The only jitter buffers that matter in your diagram are the SIP ATA and
Media Gateway. Both should have jitter buffers at the point where they
convert RTP to PCM. If adding a jitter buffer inside the network path
somewhere helps then something else is broken.
Mark
On 10/27/05, Ray Van Dolson <rayvd(a)digitalpath.net> wrote:
http://webdev.digitalpath.net/~rayvd/voice/ser_asterisk.cfg
Is my Asterisk configuration file.
Basically:
[SIP ATA] <--- 172.25.163.0/24 <http://172.25.163.0/24> IP
[Customer CPE] <--- 10.0.0.0/8 <http://10.0.0.0/8> IP
[SER] <--- 10.0.0.0/8 <http://10.0.0.0/8> IP and Public Internet IP
[Asterisk] <--- Public Internet IP
Asterisk and SER are usually far from each other.
In my setup, Asterisk normally talks to my ISP's SIP/PSTN gateway and then
passes RTP on to my ISP's media gateway. So the normal flow of RTP is:
SIP ATA -> SER -> Asterisk -> Media Gateway -> PSTN
Some of our internet connections between SER and Asterisk can be a bit
finnicky at times and introduce jitter. Since Asterisk doesn't have a
(working) jitter buffer currently, any benefits from the jitter buffer on
my
ISP's Media Gateway cannot really be realized.
So I set up the config as in my cfg above and set canreinvite to yes for
the
ATA in my sip.conf on Asterisk (also make sure your Dial() in Asterisk's
extensions.conf doesn't have a t or T in the flags list). Then the
following
happens:
1. I make a call from my SIP ATA to a PSTN number.
2. INVITE to SER
3. Normal negotiation happens. RTP is flowing as in my original flow above
(with Asterisk in the media path)
4. Asterisk now sends an INVITE to SER with an SDP payload specifying the
IP
of my ISP's media gateway. This is a public out on the Internet.
5. This requests hits the loose route portion of my route {} block.
6. I rewrite the media IP to the 10.0.0.0/8 <http://10.0.0.0/8> IP on my
SER server, but also
specify the "R" flag to "trust" the IP originially passed by
Asterisk.
This allows SER to shift it's RTP stream to the Media Gateway's IP
address.
7. I use t_on_reply(2) because I want the reply from my SIP ATA for this
re-INVITE to hit my onreply_route[2] function.
8. The re-INVITE goes out to the SIP ATA and back comes the 200 OK
response
with an SDP payload. This reply hits onreply_route[2] and I fix the media
source IP here to have the IP of the SER server.
At this point, Asterisk is now out of the media path completely!
Hope that was somewhat helpful. Don't frown too much on my messy ser.cfg
file. :-)
Ray
On Fri, Oct 28, 2005 at 11:58:22AM +1300, Patrick Jordan-Smith wrote:
Hi Ray,
Can you tell the rest of us how you did it?
Ta,
Pat
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
5:24 p.m.
When I take Asterisk out of the media path, this is correct. And I believe my
ISP's media gateway *does* have a jitter buffer.
Since Asterisk was an media endpoint before (it doesn't just proxy the rtp
on), its lack of jitter buffer was hurting us in some cases.
Ray
On Sun, Oct 30, 2005 at 08:55:13AM -0600, Mark Aiken wrote:
The only jitter buffers that matter in your diagram are the SIP ATA and
Media Gateway. Both should have jitter buffers at the point where they
convert RTP to PCM. If adding a jitter buffer inside the network path
somewhere helps then something else is broken.
Mark
9:41 p.m.
I cant see that at all from your diagram. I see only an ATA and Media
Gateway doing final conversion where jitter buffer would be useful. If
turing on a jitter buffer in Asterisk helps then one of the other 2 is
broke.
On 10/30/05, Ray Van Dolson <rayvd(a)digitalpath.net> wrote:
When I take Asterisk out of the media path, this is correct. And I believe
my
ISP's media gateway *does* have a jitter buffer.
Since Asterisk was an media endpoint before (it doesn't just proxy the rtp
on), its lack of jitter buffer was hurting us in some cases.
Ray
On Sun, Oct 30, 2005 at 08:55:13AM -0600, Mark Aiken wrote:
The only jitter buffers that matter in your diagram are the SIP ATA and
Media Gateway. Both should have jitter buffers at the point where they
convert RTP to PCM. If adding a jitter buffer inside the network path
somewhere helps then something else is broken.
Mark
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
31 Oct
31 Oct
8:31 a.m.
New subject: RES: [Serusers] Inserting SER into my voice network
I have a similar call quality problem that Ray related when using asterisk
is in the middle of media path specially, when asterisk is performing codec
transcoding. I am not sure this problem is related to asterisk lack of
jitter buffer but it seems to be.
How can we make sure asterisk is not screwing up the jitter buffer? Someone
on this list knows exactly how asterisk performs the “RTP proxing” ?
Juliano
_____
De: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] Em nome
de Mark Aiken
Enviada em: domingo, 30 de outubro de 2005 17:41
Para: Ray Van Dolson
Cc: serusers(a)lists.iptel.org
Assunto: Re: [Serusers] Inserting SER into my voice network
I cant see that at all from your diagram. I see only an ATA and Media
Gateway doing final conversion where jitter buffer would be useful. If
turing on a jitter buffer in Asterisk helps then one of the other 2 is
broke.
On 10/30/05, Ray Van Dolson <HYPERLINK
"mailto:rayvd@digitalpath.net"rayvd@digitalpath.net> wrote:
When I take Asterisk out of the media path, this is correct. And I believe
my
ISP's media gateway *does* have a jitter buffer.
Since Asterisk was an media endpoint before (it doesn't just proxy the rtp
on), its lack of jitter buffer was hurting us in some cases.
Ray
On Sun, Oct 30, 2005 at 08:55:13AM -0600, Mark Aiken wrote:
The only jitter buffers that matter in your diagram are the SIP ATA and
Media Gateway. Both should have jitter buffers at the point where they
convert RTP to PCM. If adding a jitter buffer inside the network path
somewhere helps then something else is broken.
Mark
_______________________________________________
Serusers mailing list
HYPERLINK "mailto:serusers@lists.iptel.org"serusers@lists.iptel.org
HYPERLINK
"http://lists.iptel.org/mailman/listinfo/serusers"http://lists.iptel.org/mailm
an/listinfo/serusers
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6:08 a.m.
For the case where Asterisk is handling RTP bridging, and not TDM/DAC
conversion, its not a jitter buffer you need but real-time platform
performance.
If Asterisk is the cause of jitter when its relaying media then its a flaw
in Asterisk, just like a slow/broken router will cause jitter when
forwarding from one interface to another.
The fix is not to add a jitter buffer, but quite the opposite and all
buffering and delays should be minimized.
You would need to insure that real-time priority for the entire RTP path
though Asterisk is guaranteed. I expect potential deadlock conditions would
require much work for this to be successful.
Mark
On 10/31/05, Juliano Duque da Silva <juliano.duque(a)terra.com.br> wrote:
I have a similar call quality problem that Ray related when using
asterisk is in the middle of media path specially, when asterisk is
performing codec transcoding. I am not sure this problem is related to
asterisk lack of jitter buffer but it seems to be.
How can we make sure asterisk is not screwing up the jitter buffer?
Someone on this list knows exactly how asterisk performs the "RTP proxing" ?
Juliano
------------------------------
*De:* serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] *Em
nome de *Mark Aiken
*Enviada em:* domingo, 30 de outubro de 2005 17:41
*Para:* Ray Van Dolson
*Cc:* serusers(a)lists.iptel.org
*Assunto:* Re: [Serusers] Inserting SER into my voice network
I cant see that at all from your diagram. I see only an ATA and Media
Gateway doing final conversion where jitter buffer would be useful. If
turing on a jitter buffer in Asterisk helps then one of the other 2 is
broke.
On 10/30/05, *Ray Van Dolson* <rayvd(a)digitalpath.net> wrote:
When I take Asterisk out of the media path, this is correct. And I believe
my
ISP's media gateway *does* have a jitter buffer.
Since Asterisk was an media endpoint before (it doesn't just proxy the rtp
on), its lack of jitter buffer was hurting us in some cases.
Ray
On Sun, Oct 30, 2005 at 08:55:13AM -0600, Mark Aiken wrote:
The only jitter buffers that matter in your diagram are the SIP ATA and
Media Gateway. Both should have jitter buffers at the point where they
convert RTP to PCM. If adding a jitter buffer inside the network path
somewhere helps then something else is broken.
Mark
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
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12 comments
5 participants
participants (5)
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Andres -
Juliano Duque da Silva -
Mark Aiken -
Patrick Jordan-Smith -
Ray Van Dolson