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Low, Adam

2003-Jul-28 07:29 UTC

[Asterisk-Users] RTP session traversing Asterisk server ...

I've been reading up on the SIP and related (SDP/RTP) RFC's and as I
would expect the RTP session should ideally be between the two end points of the
call, in my case the AS5300 and the 7940 which are connected on the same VLAN as
the Asterisk server.

When I sniff the packets on the VLAN I find that all RTP packets are being
relayed by the Asterisk server causing increased load on the server and
ultimately a higher latency between the two end points.

Is this a typical operation of Asterisk or is this possibly due to the fact that
some of the phones (not those used in the tests) are running NAT and Asterisk
relays all RTP packets ?

Adam


********* DISCLAIMER ********* 

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Juan Heriberto Brito Jiménez

2003-Jul-28 13:39 UTC

head link

[Asterisk-Users] RTP session traversing Asterisk server ...

Yes, i've observed the same operation :|, Adam.
I've the last CVS Asterisk, and two softphones (Linphone 1.12 and X-lite
v2 last version), both with speex code active.
When i call from one to another ... ringing ok but ... when try to talk
... the Asterisk go crazy warming "out of memory"  (i installed the
speex-dev in the server)

Are there anybody who know what's happen?

Thanxs, Heri.

PD.: Sorry my bad english :)


El lun, 28-07-2003 a las 15:29, Low, Adam escribi?:> I've been reading up on the SIP and related (SDP/RTP) RFC's and as
I would expect the RTP session should ideally be between the two end points of
the call, in my case the AS5300 and the 7940 which are connected on the same
VLAN as the Asterisk server.
> 
> When I sniff the packets on the VLAN I find that all RTP packets are being
relayed by the Asterisk server causing increased load on the server and
ultimately a higher latency between the two end points.
> 
> Is this a typical operation of Asterisk or is this possibly due to the fact
that some of the phones (not those used in the tests) are running NAT and
Asterisk relays all RTP packets ?
> 
> Adam
> 
> 
> ********* DISCLAIMER ********* 
> 
> This message and any attachment are confidential and may be privileged or
otherwise protected from disclosure and may include proprietary information. If
you are not the intended recipient, please telephone or email the sender and
delete this message and any attachment from your system. If you are not the
intended recipient you must not copy this message or attachment or disclose the
contents to any other person
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users-- 
-----------------------------------------------------------
Juan Heriberto Brito Jim?nez
Director de soluciones e-Business    
-----------------------------------------------------------
Virtual Business Europa S.L.
C/ Doctor Grau Bassas 44 (Bajo)
35007 Las Palmas de Gran Canaria
Tf:  +34 928 222960
Fax: +34 928 221521
E-mail: hbrito@virtualb.com
-----------------------------------------------------------
http://www.virtualb.com
-----------------------------------------------------------
Llave p?blica GnuPG en www.virtualb.com/keys/hbrito.asc
-----------------------------------------------------------

Dan Fernandez

2003-Jul-28 15:16 UTC

head link

[Asterisk-Users] RTP session traversing Asterisk server ...

On  your sip.conf for each sip endopoint set canreinvite = yes.

That way the rtp stream won?t go through *. The only problem though is for
ATA 186. They need canreinvite = No when they are in a NAT environment.



----- Original Message -----
From: "Low, Adam" <ALow@Prioritytelecom.com>
To: <asterisk-users@lists.digium.com>
Sent: Monday, July 28, 2003 11:29 AM
Subject: [Asterisk-Users] RTP session traversing Asterisk server ...

>
> I've been reading up on the SIP and related (SDP/RTP) RFC's and as
I wouldexpect the RTP session should ideally be between the two end points of the
call, in my case the AS5300 and the 7940 which are connected on the same
VLAN as the Asterisk server.>
> When I sniff the packets on the VLAN I find that all RTP packets are beingrelayed by the Asterisk server causing increased load on the server and
ultimately a higher latency between the two end points.>
> Is this a typical operation of Asterisk or is this possibly due to thefact that some of the phones (not those used in the tests) are running NAT
and Asterisk relays all RTP packets ?>
> Adam
>
>
> ********* DISCLAIMER *********
>
> This message and any attachment are confidential and may be privileged orotherwise protected from disclosure and may include proprietary information.
If you are not the intended recipient, please telephone or email the sender
and delete this message and any attachment from your system. If you are not
the intended recipient you must not copy this message or attachment or
disclose the contents to any other person>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Dave Packham

2003-Jul-28 19:30 UTC

head link

[Asterisk-Users] RTP session traversing Asterisk server ...

Check out this bug

http://bugs.digium.com/bug_view_page.php?bug_id=0000005

its a know problem.  I have played with the canreinvite stuff to no end and have
never gotten my Cisco Phones to do P2P RTP.  I am going to try free world dialup
to see if it does P2P with my Cisco Phones  then it might just be a message
thing on * server.

Dave Packham

>>> danfernandez00@hotmail.com 7/28/2003 4:16:16 PM >>>On  your sip.conf for each sip endopoint set canreinvite = yes.

That way the rtp stream won t go through *. The only problem though is for
ATA 186. They need canreinvite = No when they are in a NAT environment.



----- Original Message -----
From: "Low, Adam" <ALow@Prioritytelecom.com>
To: <asterisk-users@lists.digium.com>
Sent: Monday, July 28, 2003 11:29 AM
Subject: [Asterisk-Users] RTP session traversing Asterisk server ...

>
> I've been reading up on the SIP and related (SDP/RTP) RFC's and as
I wouldexpect the RTP session should ideally be between the two end points of the
call, in my case the AS5300 and the 7940 which are connected on the same
VLAN as the Asterisk server.>
> When I sniff the packets on the VLAN I find that all RTP packets are beingrelayed by the Asterisk server causing increased load on the server and
ultimately a higher latency between the two end points.>
> Is this a typical operation of Asterisk or is this possibly due to thefact that some of the phones (not those used in the tests) are running NAT
and Asterisk relays all RTP packets ?>
> Adam
>
>
> ********* DISCLAIMER *********
>
> This message and any attachment are confidential and may be privileged orotherwise protected from disclosure and may include proprietary information.
If you are not the intended recipient, please telephone or email the sender
and delete this message and any attachment from your system. If you are not
the intended recipient you must not copy this message or attachment or
disclose the contents to any other person>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com 
> http://lists.digium.com/mailman/listinfo/asterisk-users 
>_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com 
http://lists.digium.com/mailman/listinfo/asterisk-users

Low, Adam

2003-Jul-29 02:13 UTC

head link

[Asterisk-Users] RTP session traversing Asterisk server ...

Thanks all,

I spent some time on this last night with packet sniffer in hand, the
'canreinvite' option makes sense and seems to work well for me (running
latest * CVS release) when used between 79xx phones and the AS5300 gateway
although I get some somewhat expected problems with 79xx that are NAT'd
behind ADSL/cable connections.

I don't seem to be hitting the bug that Dave mentioned below ...
> -----Original Message-----
> From: Dave Packham [mailto:dave.packham@utah.edu] 
> Sent: 29 July 2003 04:30
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] RTP session traversing Asterisk 
> server ...
> 
> 
> Check out this bug
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0000005
> 
> its a know problem.  I have played with the canreinvite stuff 
> to no end and have never gotten my Cisco Phones to do P2P 
> RTP.  I am going to try free world dialup to see if it does 
> P2P with my Cisco Phones  then it might just be a message 
> thing on * server.
> 
> Dave Packham
> 
> 
> >>> danfernandez00@hotmail.com 7/28/2003 4:16:16 PM >>>
> On  your sip.conf for each sip endopoint set canreinvite = yes.
> 
> That way the rtp stream won t go through *. The only problem 
> though is for
> ATA 186. They need canreinvite = No when they are in a NAT 
> environment.
> 
> 
> 
> ----- Original Message -----
> From: "Low, Adam" <ALow@Prioritytelecom.com>
> To: <asterisk-users@lists.digium.com>
> Sent: Monday, July 28, 2003 11:29 AM
> Subject: [Asterisk-Users] RTP session traversing Asterisk server ...
> 
> 
> >
> > I've been reading up on the SIP and related (SDP/RTP) RFC's 
> and as I would
> expect the RTP session should ideally be between the two end 
> points of the
> call, in my case the AS5300 and the 7940 which are connected 
> on the same
> VLAN as the Asterisk server.
> >
> > When I sniff the packets on the VLAN I find that all RTP 
> packets are being
> relayed by the Asterisk server causing increased load on the 
> server and
> ultimately a higher latency between the two end points.
> >
> > Is this a typical operation of Asterisk or is this possibly 
> due to the
> fact that some of the phones (not those used in the tests) 
> are running NAT
> and Asterisk relays all RTP packets ?
> >
> > Adam
> >
> >
> > ********* DISCLAIMER *********
> >
> > This message and any attachment are confidential and may be 
> privileged or
> otherwise protected from disclosure and may include 
> proprietary information.
> If you are not the intended recipient, please telephone or 
> email the sender
> and delete this message and any attachment from your system. 
> If you are not
> the intended recipient you must not copy this message or attachment or
> disclose the contents to any other person
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com 
> > http://lists.digium.com/mailman/listinfo/asterisk-users 
> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com 
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 

********* DISCLAIMER ********* 

This message and any attachment are confidential and may be privileged or
otherwise protected from disclosure and may include proprietary information. If
you are not the intended recipient, please telephone or email the sender and
delete this message and any attachment from your system. If you are not the
intended recipient you must not copy this message or attachment or disclose the
contents to any other person

Dave Packham

2003-Jul-29 06:43 UTC

head link

[Asterisk-Users] RTP session traversing Asterisk server ...

can you share the SIP conf entries that you are using to get this to work?   I
have played with the canreinvite and reinvite entries but cannot make my
7960's do P2P  I am running the 5.1 SIP code on the phones.

Dave

>>> ALow@Prioritytelecom.com 7/29/2003 3:13:54 AM >>>Thanks all,

I spent some time on this last night with packet sniffer in hand, the
'canreinvite' option makes sense and seems to work well for me (running
latest * CVS release) when used between 79xx phones and the AS5300 gateway
although I get some somewhat expected problems with 79xx that are NAT'd
behind ADSL/cable connections.

I don't seem to be hitting the bug that Dave mentioned below ...
> -----Original Message-----
> From: Dave Packham [mailto:dave.packham@utah.edu] 
> Sent: 29 July 2003 04:30
> To: asterisk-users@lists.digium.com 
> Subject: Re: [Asterisk-Users] RTP session traversing Asterisk 
> server ...
> 
> 
> Check out this bug
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0000005 
> 
> its a know problem.  I have played with the canreinvite stuff 
> to no end and have never gotten my Cisco Phones to do P2P 
> RTP.  I am going to try free world dialup to see if it does 
> P2P with my Cisco Phones  then it might just be a message 
> thing on * server.
> 
> Dave Packham
> 
> 
> >>> danfernandez00@hotmail.com 7/28/2003 4:16:16 PM >>>
> On  your sip.conf for each sip endopoint set canreinvite = yes.
> 
> That way the rtp stream won t go through *. The only problem 
> though is for
> ATA 186. They need canreinvite = No when they are in a NAT 
> environment.
> 
> 
> 
> ----- Original Message -----
> From: "Low, Adam" <ALow@Prioritytelecom.com>
> To: <asterisk-users@lists.digium.com>
> Sent: Monday, July 28, 2003 11:29 AM
> Subject: [Asterisk-Users] RTP session traversing Asterisk server ...
> 
> 
> >
> > I've been reading up on the SIP and related (SDP/RTP) RFC's 
> and as I would
> expect the RTP session should ideally be between the two end 
> points of the
> call, in my case the AS5300 and the 7940 which are connected 
> on the same
> VLAN as the Asterisk server.
> >
> > When I sniff the packets on the VLAN I find that all RTP 
> packets are being
> relayed by the Asterisk server causing increased load on the 
> server and
> ultimately a higher latency between the two end points.
> >
> > Is this a typical operation of Asterisk or is this possibly 
> due to the
> fact that some of the phones (not those used in the tests) 
> are running NAT
> and Asterisk relays all RTP packets ?
> >
> > Adam
> >
> >
> > ********* DISCLAIMER *********
> >
> > This message and any attachment are confidential and may be 
> privileged or
> otherwise protected from disclosure and may include 
> proprietary information.
> If you are not the intended recipient, please telephone or 
> email the sender
> and delete this message and any attachment from your system. 
> If you are not
> the intended recipient you must not copy this message or attachment or
> disclose the contents to any other person
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com 
> > http://lists.digium.com/mailman/listinfo/asterisk-users 
> >
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com 
> http://lists.digium.com/mailman/listinfo/asterisk-users 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com 
> http://lists.digium.com/mailman/listinfo/asterisk-users 
> 

********* DISCLAIMER ********* 

This message and any attachment are confidential and may be privileged or
otherwise protected from disclosure and may include proprietary information. If
you are not the intended recipient, please telephone or email the sender and
delete this message and any attachment from your system. If you are not the
intended recipient you must not copy this message or attachment or disclose the
contents to any other person


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com 
http://lists.digium.com/mailman/listinfo/asterisk-users

Low, Adam

2003-Jul-29 07:01 UTC

head link

[Asterisk-Users] RTP session traversing Asterisk server ...

Sure, nothing special though:

[4840]
type=friend
username=4840
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband

[4842]
type=friend
username=4842
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband


> -----Original Message-----
> From: Dave Packham [mailto:dave.packham@utah.edu] 
> Sent: 29 July 2003 15:43
> To: asterisk-users@lists.digium.com; ALow@Prioritytelecom.com
> Subject: RE: [Asterisk-Users] RTP session traversing Asterisk 
> server ...
> 
> 
> can you share the SIP conf entries that you are using to get 
> this to work?   I have played with the canreinvite and 
> reinvite entries but cannot make my 7960's do P2P  I am 
> running the 5.1 SIP code on the phones.   
> 
> Dave
> 
> 
> >>> ALow@Prioritytelecom.com 7/29/2003 3:13:54 AM >>>
> Thanks all,
> 
> I spent some time on this last night with packet sniffer in 
> hand, the 'canreinvite' option makes sense and seems to work 
> well for me (running latest * CVS release) when used between 
> 79xx phones and the AS5300 gateway although I get some 
> somewhat expected problems with 79xx that are NAT'd behind 
> ADSL/cable connections.
> 
> I don't seem to be hitting the bug that Dave mentioned below ...
> 
> > -----Original Message-----
> > From: Dave Packham [mailto:dave.packham@utah.edu] 
> > Sent: 29 July 2003 04:30
> > To: asterisk-users@lists.digium.com 
> > Subject: Re: [Asterisk-Users] RTP session traversing Asterisk 
> > server ...
> > 
> > 
> > Check out this bug
> > 
> > http://bugs.digium.com/bug_view_page.php?bug_id=0000005 
> > 
> > its a know problem.  I have played with the canreinvite stuff 
> > to no end and have never gotten my Cisco Phones to do P2P 
> > RTP.  I am going to try free world dialup to see if it does 
> > P2P with my Cisco Phones  then it might just be a message 
> > thing on * server.
> > 
> > Dave Packham
> > 
> > 
> > >>> danfernandez00@hotmail.com 7/28/2003 4:16:16 PM
>>>
> > On  your sip.conf for each sip endopoint set canreinvite = yes.
> > 
> > That way the rtp stream won t go through *. The only problem 
> > though is for
> > ATA 186. They need canreinvite = No when they are in a NAT 
> > environment.
> > 
> > 
> > 
> > ----- Original Message -----
> > From: "Low, Adam" <ALow@Prioritytelecom.com>
> > To: <asterisk-users@lists.digium.com>
> > Sent: Monday, July 28, 2003 11:29 AM
> > Subject: [Asterisk-Users] RTP session traversing Asterisk server ...
> > 
> > 
> > >
> > > I've been reading up on the SIP and related (SDP/RTP)
RFC's
> > and as I would
> > expect the RTP session should ideally be between the two end 
> > points of the
> > call, in my case the AS5300 and the 7940 which are connected 
> > on the same
> > VLAN as the Asterisk server.
> > >
> > > When I sniff the packets on the VLAN I find that all RTP 
> > packets are being
> > relayed by the Asterisk server causing increased load on the 
> > server and
> > ultimately a higher latency between the two end points.
> > >
> > > Is this a typical operation of Asterisk or is this possibly 
> > due to the
> > fact that some of the phones (not those used in the tests) 
> > are running NAT
> > and Asterisk relays all RTP packets ?
> > >
> > > Adam
> > >
> > >
> > > ********* DISCLAIMER *********
> > >
> > > This message and any attachment are confidential and may be 
> > privileged or
> > otherwise protected from disclosure and may include 
> > proprietary information.
> > If you are not the intended recipient, please telephone or 
> > email the sender
> > and delete this message and any attachment from your system. 
> > If you are not
> > the intended recipient you must not copy this message or 
> attachment or
> > disclose the contents to any other person
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com 
> > > http://lists.digium.com/mailman/listinfo/asterisk-users 
> > >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com 
> > http://lists.digium.com/mailman/listinfo/asterisk-users 
> > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com 
> > http://lists.digium.com/mailman/listinfo/asterisk-users 
> > 
> 
> 
> ********* DISCLAIMER ********* 
> 
> This message and any attachment are confidential and may be 
> privileged or otherwise protected from disclosure and may 
> include proprietary information. If you are not the intended 
> recipient, please telephone or email the sender and delete 
> this message and any attachment from your system. If you are 
> not the intended recipient you must not copy this message or 
> attachment or disclose the contents to any other person 
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com 
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 

********* DISCLAIMER ********* 

This message and any attachment are confidential and may be privileged or
otherwise protected from disclosure and may include proprietary information. If
you are not the intended recipient, please telephone or email the sender and
delete this message and any attachment from your system. If you are not the
intended recipient you must not copy this message or attachment or disclose the
contents to any other person

Dave Packham

2003-Jul-29 11:08 UTC

head link

[Asterisk-Users] RTP session traversing Asterisk server ...

made those changes and still no P2P

[70900]
type=friend
insecure=yes
username=70900
secret=youwish
host=dynamic
context = campus
mailbox=70900
canreinvite=yes
nat=no
qualify=200
dtmfmode=inband

is what I have for my Cisco 7960's

Dave
>>> ALow@Prioritytelecom.com 7/29/2003 8:01:41 AM >>>Sure, nothing special though:

[4840]
type=friend
username=4840
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband

[4842]
type=friend
username=4842
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband


> -----Original Message-----
> From: Dave Packham [mailto:dave.packham@utah.edu] 
> Sent: 29 July 2003 15:43
> To: asterisk-users@lists.digium.com; ALow@Prioritytelecom.com 
> Subject: RE: [Asterisk-Users] RTP session traversing Asterisk 
> server ...
> 
> 
> can you share the SIP conf entries that you are using to get 
> this to work?   I have played with the canreinvite and 
> reinvite entries but cannot make my 7960's do P2P  I am 
> running the 5.1 SIP code on the phones.   
> 
> Dave
> 
> 
> >>> ALow@Prioritytelecom.com 7/29/2003 3:13:54 AM >>>
> Thanks all,
> 
> I spent some time on this last night with packet sniffer in 
> hand, the 'canreinvite' option makes sense and seems to work 
> well for me (running latest * CVS release) when used between 
> 79xx phones and the AS5300 gateway although I get some 
> somewhat expected problems with 79xx that are NAT'd behind 
> ADSL/cable connections.
> 
> I don't seem to be hitting the bug that Dave mentioned below ...
> 
> > -----Original Message-----
> > From: Dave Packham [mailto:dave.packham@utah.edu] 
> > Sent: 29 July 2003 04:30
> > To: asterisk-users@lists.digium.com 
> > Subject: Re: [Asterisk-Users] RTP session traversing Asterisk 
> > server ...
> > 
> > 
> > Check out this bug
> > 
> > http://bugs.digium.com/bug_view_page.php?bug_id=0000005 
> > 
> > its a know problem.  I have played with the canreinvite stuff 
> > to no end and have never gotten my Cisco Phones to do P2P 
> > RTP.  I am going to try free world dialup to see if it does 
> > P2P with my Cisco Phones  then it might just be a message 
> > thing on * server.
> > 
> > Dave Packham
> > 
> > 
> > >>> danfernandez00@hotmail.com 7/28/2003 4:16:16 PM
>>>
> > On  your sip.conf for each sip endopoint set canreinvite = yes.
> > 
> > That way the rtp stream won t go through *. The only problem 
> > though is for
> > ATA 186. They need canreinvite = No when they are in a NAT 
> > environment.
> > 
> > 
> > 
> > ----- Original Message -----
> > From: "Low, Adam" <ALow@Prioritytelecom.com>
> > To: <asterisk-users@lists.digium.com>
> > Sent: Monday, July 28, 2003 11:29 AM
> > Subject: [Asterisk-Users] RTP session traversing Asterisk server ...
> > 
> > 
> > >
> > > I've been reading up on the SIP and related (SDP/RTP)
RFC's
> > and as I would
> > expect the RTP session should ideally be between the two end 
> > points of the
> > call, in my case the AS5300 and the 7940 which are connected 
> > on the same
> > VLAN as the Asterisk server.
> > >
> > > When I sniff the packets on the VLAN I find that all RTP 
> > packets are being
> > relayed by the Asterisk server causing increased load on the 
> > server and
> > ultimately a higher latency between the two end points.
> > >
> > > Is this a typical operation of Asterisk or is this possibly 
> > due to the
> > fact that some of the phones (not those used in the tests) 
> > are running NAT
> > and Asterisk relays all RTP packets ?
> > >
> > > Adam
> > >
> > >
> > > ********* DISCLAIMER *********
> > >
> > > This message and any attachment are confidential and may be 
> > privileged or
> > otherwise protected from disclosure and may include 
> > proprietary information.
> > If you are not the intended recipient, please telephone or 
> > email the sender
> > and delete this message and any attachment from your system. 
> > If you are not
> > the intended recipient you must not copy this message or 
> attachment or
> > disclose the contents to any other person
> > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com 
> > > http://lists.digium.com/mailman/listinfo/asterisk-users 
> > >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com 
> > http://lists.digium.com/mailman/listinfo/asterisk-users 
> > 
> > _______________________________________________
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Dave Packham

2003-Jul-29 13:00 UTC

head link

[Asterisk-Users] RTP session traversing Asterisk server ...

OK calls thru the * server are looped and calls with the same phones thru Free
WOrld Dialup are P2P.....  same configs...

Anyone have any ideas?  I know its a bug but we need to fix this one.... I think
its pretty big one.  it would HAMMER the scalability of * servers

Dave
>>> ALow@Prioritytelecom.com 7/29/2003 8:01:41 AM >>>Sure, nothing special though:

[4840]
type=friend
username=4840
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband

[4842]
type=friend
username=4842
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband


> -----Original Message-----
> From: Dave Packham [mailto:dave.packham@utah.edu] 
> Sent: 29 July 2003 15:43
> To: asterisk-users@lists.digium.com; ALow@Prioritytelecom.com 
> Subject: RE: [Asterisk-Users] RTP session traversing Asterisk 
> server ...
> 
> 
> can you share the SIP conf entries that you are using to get 
> this to work?   I have played with the canreinvite and 
> reinvite entries but cannot make my 7960's do P2P  I am 
> running the 5.1 SIP code on the phones.   
> 
> Dave
> 
> 
> >>> ALow@Prioritytelecom.com 7/29/2003 3:13:54 AM >>>
> Thanks all,
> 
> I spent some time on this last night with packet sniffer in 
> hand, the 'canreinvite' option makes sense and seems to work 
> well for me (running latest * CVS release) when used between 
> 79xx phones and the AS5300 gateway although I get some 
> somewhat expected problems with 79xx that are NAT'd behind 
> ADSL/cable connections.
> 
> I don't seem to be hitting the bug that Dave mentioned below ...
> 
> > -----Original Message-----
> > From: Dave Packham [mailto:dave.packham@utah.edu] 
> > Sent: 29 July 2003 04:30
> > To: asterisk-users@lists.digium.com 
> > Subject: Re: [Asterisk-Users] RTP session traversing Asterisk 
> > server ...
> > 
> > 
> > Check out this bug
> > 
> > http://bugs.digium.com/bug_view_page.php?bug_id=0000005 
> > 
> > its a know problem.  I have played with the canreinvite stuff 
> > to no end and have never gotten my Cisco Phones to do P2P 
> > RTP.  I am going to try free world dialup to see if it does 
> > P2P with my Cisco Phones  then it might just be a message 
> > thing on * server.
> > 
> > Dave Packham
> > 
> > 
> > >>> danfernandez00@hotmail.com 7/28/2003 4:16:16 PM
>>>
> > On  your sip.conf for each sip endopoint set canreinvite = yes.
> > 
> > That way the rtp stream won t go through *. The only problem 
> > though is for
> > ATA 186. They need canreinvite = No when they are in a NAT 
> > environment.
> > 
> > 
> > 
> > ----- Original Message -----
> > From: "Low, Adam" <ALow@Prioritytelecom.com>
> > To: <asterisk-users@lists.digium.com>
> > Sent: Monday, July 28, 2003 11:29 AM
> > Subject: [Asterisk-Users] RTP session traversing Asterisk server ...
> > 
> > 
> > >
> > > I've been reading up on the SIP and related (SDP/RTP)
RFC's
> > and as I would
> > expect the RTP session should ideally be between the two end 
> > points of the
> > call, in my case the AS5300 and the 7940 which are connected 
> > on the same
> > VLAN as the Asterisk server.
> > >
> > > When I sniff the packets on the VLAN I find that all RTP 
> > packets are being
> > relayed by the Asterisk server causing increased load on the 
> > server and
> > ultimately a higher latency between the two end points.
> > >
> > > Is this a typical operation of Asterisk or is this possibly 
> > due to the
> > fact that some of the phones (not those used in the tests) 
> > are running NAT
> > and Asterisk relays all RTP packets ?
> > >
> > > Adam
> > >
> > >
> > > ********* DISCLAIMER *********
> > >
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> > >
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you are not the intended recipient, please telephone or email the sender and
delete this message and any attachment from your system. If you are not the
intended recipient you must not copy this message or attachment or disclose the
contents to any other person


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