Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!
On Sat, 15 Aug 2015 12:42:38 -0300 Joshua Colp wrote: > > I am not sure why this hasn't bit anyone else. Perhaps most > > Asterisk systems are in one of two classes, connecting to all NAT > > phones or connecting to all public phones, and I am in a minority > > situation where I am talking to a mix of setups. > > Most people run without direct media unless they know the network > topology will allow it 100%. Perhaps but the default is to run it. Perhaps the default should be "no" to prevent these problems. On the other hand, the documentation seemed to suggest that the default should have worked anyway. One leg was public, the other behind a NAT. It should recognize the latter and not try to put then in direct contact. It's almost like it saw the public one and didn't bother checking the other. Or, it checked both with an OR instead of an AND as I said. That seems more likely since it didn't matter who started the call. I don't really care at this point. If 1% of the calls go through the server when they didn't really need to it's no big deal. Cheers. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!
On Sat, Aug 15, 2015, at 12:08 PM, D'Arcy J.M. Cain wrote: > On Sat, 15 Aug 2015 16:30:39 +0800 > Michael Dupree wrote: > > Not 100% ure, but maybe play with the canreinvite or directmedia > > settings. > > Yes! That was it. Just for future searches here is what I did. I > added "directmedia = no" in sip.conf. This fixed the issue. > > I believe that Asterisk was getting confused when one leg was inside > NAT and the other was outside. Perhaps there was an "OR" where there > should be an "AND". It makes sense because the other user was the one > outside NAT and he could hear me and I could not hear him no matter who > initiated the call. He could make outside calls because both he and my > provider were on public IPs. > > I am not sure why this hasn't bit anyone else. Perhaps most Asterisk > systems are in one of two classes, connecting to all NAT phones or > connecting to all public phones, and I am in a minority situation where > I am talking to a mix of setups. Most people run without direct media unless they know the network topology will allow it 100%. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!
On Sat, 15 Aug 2015 16:30:39 +0800 Michael Dupree wrote: > Not 100% ure, but maybe play with the canreinvite or directmedia > settings. Yes! That was it. Just for future searches here is what I did. I added "directmedia = no" in sip.conf. This fixed the issue. I believe that Asterisk was getting confused when one leg was inside NAT and the other was outside. Perhaps there was an "OR" where there should be an "AND". It makes sense because the other user was the one outside NAT and he could hear me and I could not hear him no matter who initiated the call. He could make outside calls because both he and my provider were on public IPs. I am not sure why this hasn't bit anyone else. Perhaps most Asterisk systems are in one of two classes, connecting to all NAT phones or connecting to all public phones, and I am in a minority situation where I am talking to a mix of setups. Thanks for that. I was going nuts trying to figure this out. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio - doesn't seem to be NAT issue
Not 100% ure, but maybe play with the canreinvite or directmedia settings. On Wed, Aug 12, 2015 at 3:10 AM, D'Arcy J.M. Cain wrote: > I have been banging my head against the wall for weeks now on this > one. I have a switch running NetBSD and Asterisk 11.19.0 although I > have had this problem on older versions as well. I, and my users, can > call out, we can receive calls, quality is excellent but I cannot talk > with one user. The different elements are as follows: > > The switch as described above which is in a server room on the Internet > backbone with a public IP address. > > My home system which is behind a bridged modem through a Linksys > WRT54GS with priority given to my ATA. The ATA is a Cisco SPA112. I > also have an actual SIP phone. The problem happens with both. > Obviously I am using NAT but both devices work just fine if I am going > to the PSTN. > > My user who is also going through a bridged modem to a Linksys SPA-2102 > which is doing the PPPOE so it has a public IP address and no NAT > involved although it serves NAT for the connected computer. > > So here is the problem. While both of us have no problems externally, > when we call each other we get one way audio and it is always from me > to him no matter who initiates the call. > > A further test, I can call from the SIP phone to the ATA connected > phone and vice versa just fine. That involves two devices behind the > same NAT but since they still need to use the server as an intermediary > I can't see how that would matter. > > Given that both of us can make and accept calls and the server is > simply connecting two separate channels I can't see where the problem > might lie. Can anyone suggest a possible setup issue? > > I have tried so many things but I am willing to try them again. Feel > free to make any suggestion no matter how silly. I really need to fix > this. > > Cheers. > > > -- > D'Arcy J.M. Cain > System Administrator, Vex.Net > http://www.Vex.Net/ IM:da...@vex.net > VoIP: sip:da...@vex.net > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Michael Dupree jr. p: +1-248-935-4147 f: +1-866-671-6867 Skype: MichaelDupreeJr PGP Pub Key: http://www.michaeldupree.net/?page_id=53 This is a private message. This e-mail message, and any attachments thereto, is for the sole use of the intended recipient(s) and may contain legally privileged and/or confidential information. Any unauthorized review, use, disclosure or distribution is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and permanently delete all copies of the original message. --- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio - doesn't seem to be NAT issue
Hi D'Arcy >> that the server IP for RTP as specified in the initial SIP is correct? >Both the server and client are outside of NAT so I don't know what this might mean. They both have public IPs. This was a problem we had when the RTP server negotiated in SIP with our VOIP ITSP on one side of the connection, differed from the IP we were expecting on that side of the connection and was blocked in our firewall. Once we perused the SIP traffic we noted this and added the extra IP to the firewall for RTP traffic. >> We had slightly different parameters, e. g. that we would have no RTP >> at all, but a call that did connect to total silence, dialed from >> either side. >Was NAT involved? Yes, NAT was being done at both ends, but it turned out that NATing was not the problem. >> Also check what RTP port ranges are being used - I have had this >> one-directional problem where the port range in /etc/asterisk/rtp.conf >> was too broad, and the firewall on my server was only allowing a >> smaller subset of RTP ports. >rtpstart=1 >rtpend=2 >which is exactly what my packet filter allows through. I assume you have tried turning your packet filter or firewall off completely (just for a moment) to see if it helped? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One way audio - doesn't seem to be NAT issue
On Thu, 13 Aug 2015 10:41:31 +0200 "Stefan Viljoen" wrote: > Have you checked your RTP port ranges (I'm sure you have), and also Yes. The ATA is using a range well within the range open on the server. > that the server IP for RTP as specified in the initial SIP is correct? Both the server and client are outside of NAT so I don't know what this might mean. They both have public IPs. > Not sure how this will relate to your setup, but we had something > similar here using Asterisk 1.8.11.0 on both sides of the connection, > via a VOIP service provider in the middle. This is an Asterisk server talking to an ATA. > We had slightly different parameters, e. g. that we would have no RTP > at all, but a call that did connect to total silence, dialed from > either side. Was NAT involved? > Also check what RTP port ranges are being used - I have had this > one-directional problem where the port range > in /etc/asterisk/rtp.conf was too broad, and the firewall on my > server was only allowing a smaller subset of RTP ports. rtpstart=1 rtpend=2 which is exactly what my packet filter allows through. > It might require some careful tracing of SIP messages, maybe you can > try this? Specifically try to determine what RTP port number is being > negotiated when you have your zero-audio back from the remote party - > what RTP port and RTP server IP is he using at that moment on his > side? I will check that. Thanks for your suggestions. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio - doesn't seem to be NAT issue
Hi D'arcy
Have you checked your RTP port ranges (I'm sure you have), and also that the
server IP for RTP as specified in the initial SIP is correct?
Not sure how this will relate to your setup, but we had something similar
here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP
service provider in the middle.
We had slightly different parameters, e. g. that we would have no RTP at
all, but a call that did connect to total silence, dialed from either side.
We subscribe to two trunk numbers provided by the VOIP service provider at
each site in Asterisk.
It turned out after carefully looking at the SIP flowing back and forth that
the service provider was providing an RTP server IP that specified not the
same IP as the SIP server (which is their standard practice) but a
-different- RTP server IP.
Due to the routing we have, neither system on either side of the SIP
negotiated call could send packets to this "new" RTP server IP.
We therefore added a route that specifically allowed that "new" RTP server
IP to be reached by both machines on both sides of the VOIP service provider
link.
So can you carefully check that the SIP-negotiated RTP streams are going to
IPs that are reachable in BOTH directions?
Also check what RTP port ranges are being used - I have had this
one-directional problem where the port range in /etc/asterisk/rtp.conf was
too broad, and the firewall on my server was only allowing a smaller subset
of RTP ports.
E. g. /etc/asterisk/rtp.conf specified 1 - 5 as allowable RTP ports,
but my firewalld firewall under Centos was only allowing 1 - 2 - so
I'd regularly get that my SECOND call to test the server would have audio in
one direction - because
Asterisk was allocating an RTP port on one side of the SIP call that was
outside the range my firewalld was allowing.
It might require some careful tracing of SIP messages, maybe you can try
this? Specifically try to determine what RTP port number is being negotiated
when you have your zero-audio back from the remote party - what RTP port and
RTP server IP is he using at that moment on his side?
Is that port allowed through all the PPP / network segments between you? Is
the IP / IPs between you used to transfer RTP reachable from his side?
Message: 1
Date: Tue, 11 Aug 2015 15:10:44 -0400
From: "D'Arcy J.M. Cain"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users] One way audio - doesn't seem to be NAT issue
Message-ID: <20150811151044.79872ce9@imp>
Content-Type: text/plain; charset=US-ASCII
Given that both of us can make and accept calls and the server is simply
connecting two separate channels I can't see where the problem might lie.
Can anyone suggest a possible setup issue?
I have tried so many things but I am willing to try them again. Feel free
to make any suggestion no matter how silly. I really need to fix this.
Cheers.
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] One way audio - doesn't seem to be NAT issue
On Tue, Aug 11, 2015, at 04:10 PM, D'Arcy J.M. Cain wrote: > I have been banging my head against the wall for weeks now on this > one. I have a switch running NetBSD and Asterisk 11.19.0 although I > have had this problem on older versions as well. I, and my users, can > call out, we can receive calls, quality is excellent but I cannot talk > with one user. The different elements are as follows: I'd suggest getting a packet capture to see the RTP traffic to see the actual path of things, not just thinking of what it should be. Media doesn't just get lost. It's told to go somewhere ultimately and either that is incorrect for some reason or something is blocking it. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One way audio - doesn't seem to be NAT issue
I have been banging my head against the wall for weeks now on this one. I have a switch running NetBSD and Asterisk 11.19.0 although I have had this problem on older versions as well. I, and my users, can call out, we can receive calls, quality is excellent but I cannot talk with one user. The different elements are as follows: The switch as described above which is in a server room on the Internet backbone with a public IP address. My home system which is behind a bridged modem through a Linksys WRT54GS with priority given to my ATA. The ATA is a Cisco SPA112. I also have an actual SIP phone. The problem happens with both. Obviously I am using NAT but both devices work just fine if I am going to the PSTN. My user who is also going through a bridged modem to a Linksys SPA-2102 which is doing the PPPOE so it has a public IP address and no NAT involved although it serves NAT for the connected computer. So here is the problem. While both of us have no problems externally, when we call each other we get one way audio and it is always from me to him no matter who initiates the call. A further test, I can call from the SIP phone to the ATA connected phone and vice versa just fine. That involves two devices behind the same NAT but since they still need to use the server as an intermediary I can't see how that would matter. Given that both of us can make and accept calls and the server is simply connecting two separate channels I can't see where the problem might lie. Can anyone suggest a possible setup issue? I have tried so many things but I am willing to try them again. Feel free to make any suggestion no matter how silly. I really need to fix this. Cheers. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
