RE: [Asterisk-Users] RTP session traversing Asterisk server ...
Dear all, Concerning the REINVITE discussion... Does anybody can confirm if the Cisco ATA186 accepts the REINVITE Sip Message, and if is compatible with this feature? In other words, is it possible to make the ATA186 change the RTP destination and start sending the media packets straight to destination point instead of keep sending to Asterisk? Thanks Regards, Artur C. Severo Eng., M.Sc. Network Engineer Tel: 55 51 3328 0636 #242 Low, Adam wrote: 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 ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP session traversing Asterisk server...
Ricardo, You are right about the contact field in the INVITE message. It does display the address or our Asterisk proxy. It seems to me that this field is used for endpoints to exchange future SIP messages among themselves and not to set up the RTP stream. I have found that the SDP Connection (c) field in the invite also reflects the IP of the Asterisk box after the message leaves the proxy. The 200 OK reflects the same symptoms. I think that this is the reason the RTP stream is being set up between the endpoint and the server. Do you think that the contact field and connection field being incorrect may be related? You also have mentioned that you have not seen a way to configure this with Asterisk. How about other SIP proxies such as VOCAL? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 2:23 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server... Dave, You can use a sniffer to view the contact field in the INVITE Message that the Originating Phone sends to *. Then look at the INVITE Message that * sends to the remote phone and compare the contact filed. You will see that the IP Address is changed to reflect the IP of *. If you want pure P2P then that address needs to remain the same. I have not seen how you can do that with *. Ricardo - Original Message - From: Dave Packham [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 3:00 PM Subject: RE: [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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED] Sent: 29 July 2003 15:43 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED] Sent: 29 July 2003 04:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server ... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] To: [EMAIL PROTECTED] 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
Re: [Asterisk-Users] RTP session traversing Asterisk server...
Hi Andrew, After looking at some SIP messages again I too think the (c) field in the SDP is what determines the RTP endpoints. It's just that in our case it is always the same as the Contact field. In any case what you see here is that * is making some changes here to make sure SIP messages and RTP stream passes through it. If what you want is a plain but powerful SIP Proxy then take a look at (SER) http://www.iptel.org. That is what we use to run our SIP P2P network. We only use * for our PBX. Regards, Ricardo http://www.telesip.net - Original Message - From: Andrew Reich [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 01, 2003 4:20 PM Subject: RE: [Asterisk-Users] RTP session traversing Asterisk server... Ricardo, You are right about the contact field in the INVITE message. It does display the address or our Asterisk proxy. It seems to me that this field is used for endpoints to exchange future SIP messages among themselves and not to set up the RTP stream. I have found that the SDP Connection (c) field in the invite also reflects the IP of the Asterisk box after the message leaves the proxy. The 200 OK reflects the same symptoms. I think that this is the reason the RTP stream is being set up between the endpoint and the server. Do you think that the contact field and connection field being incorrect may be related? You also have mentioned that you have not seen a way to configure this with Asterisk. How about other SIP proxies such as VOCAL? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 2:23 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server... Dave, You can use a sniffer to view the contact field in the INVITE Message that the Originating Phone sends to *. Then look at the INVITE Message that * sends to the remote phone and compare the contact filed. You will see that the IP Address is changed to reflect the IP of *. If you want pure P2P then that address needs to remain the same. I have not seen how you can do that with *. Ricardo - Original Message - From: Dave Packham [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 3:00 PM Subject: RE: [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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED] Sent: 29 July 2003 15:43 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED] Sent: 29 July 2003 04:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server ... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED
Re: [Asterisk-Users] RTP session traversing Asterisk server...
Hi, Cisco 7940/60 does P2P with FWD. BR, Dan - Original Message - From: Dave Packham [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 5:30 AM Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [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:[EMAIL PROTECTED] Sent: 29 July 2003 04:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server ... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [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:[EMAIL PROTECTED] Sent: 29 July 2003 15:43 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED] Sent: 29 July 2003 04:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server ... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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
RE: [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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED] Sent: 29 July 2003 15:43 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED] Sent: 29 July 2003 04:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server ... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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
RE: [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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED] Sent: 29 July 2003 15:43 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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:[EMAIL PROTECTED] Sent: 29 July 2003 04:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server ... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged
Re: [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 escribió: 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Juan Heriberto Brito Jiménez 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: [EMAIL PROTECTED] --- http://www.virtualb.com --- Llave pública GnuPG en www.virtualb.com/keys/hbrito.asc --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [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 [EMAIL PROTECTED] To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP session traversing Asterisk server...
Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
