Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such.... * nat (yes): No problem here either.... * defaultuser (1003 at example.com): Does the "@example.com" have to point to the UA (i.e., (1003 at ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing "host=dynamic" takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is "host=dynamic" sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003 at ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick.
Danny Nicholas
2013-Jan-03 17:51 UTC
[asterisk-users] Moving User Agent To Remote Location
-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, January 03, 2013 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Moving User Agent To Remote Location Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such.... * nat (yes): No problem here either.... * defaultuser (1003 at example.com): Does the "@example.com" have to point to the UA (i.e., (1003 at ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing "host=dynamic" takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is "host=dynamic" sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003 at ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick. I'm going to vote for the RTP issue. If you are establishing a call but have no audio, you are getting the 5060 port, but not the 10000-20000 range that RTP normally expects. A "better" practice is to allocate 4 ports per line you expect to use in rtp.conf (10000-20000 would allow 2500 lines; much more that most folks need and more "holes" to monitor).
Stelios Koroneos
2013-Jan-03 20:34 UTC
[asterisk-users] Moving User Agent To Remote Location
Assuming this is not a NAT issue (which looks like it), the "no audio" when signaling works usually means some misconfiguration of codecs Stelios
On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote:> Hello Everyone, > > Before getting into SIP and RTP traces, I wanted to clarify some of > the sip.conf settings that may to some seem redundant or have a > misconception with. I do apologize if this has been discussed time and > time again as I would imagine. If anything, this email would make > google search results that much stronger :). > > With the UA local to my network I had tested two way audio, and now > with the phone outside of network, we have no way audio. Before > discussing NAT (which is enabled on the peer), and port forwarding > (which is setup on the remote location), I would like to make sure I > fully understand all the sip.conf settings. We are using Asterisk > realtime via sip_buddies, and the fields in question are: > > (Enclosed in brackets are an example value for the setting) > > * host (dynamic): No problem here. Just wanted to mention that it's > set as such.... > * nat (yes): No problem here either.... > * defaultuser (1003 at example.com): Does the "@example.com" have to > point to the UA (i.e., (1003 at ua-public-ip), or is it just a name type > field? > * fullcontact: What to put here for a UA that is running at a remote > location with dynamic external IP? > * ipaddr (ua-public-ip): I did try setting it to the public ip of the > UA, but is that really practical? > What if I don't know where the initial registration request is coming > from? I am guessing "host=dynamic" takes care of that. > * defaultip?? > * dynamic: Should this be set to yes, or is "host=dynamic" sufficient? > > The phone registers fine, and terminates a call through our providers. > Just no audio both ways, which would suggest something more that an > RTP issue which should at least have one way outgoing audio. > > Things that have been attempted: > * Port forwarding to the phone > * Changing defaultuser to 1003 at ua-public-ip. This made our OpenSIPS > sip proxy through a fit. > > Things I will attempt today: > Calling the UA extension from an extension here > SIP trace > > Your help is greatly appreciated!!! > > Nick. >Hi Is your directmedia/canreinvite (depending on version) set to no? Regards Ish -- Ishfaq Malik <ish at pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552
On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote:> Hello Everyone, > > Before getting into SIP and RTP traces, I wanted to clarify some of > the sip.conf settings that may to some seem redundant or have a > misconception with. I do apologize if this has been discussed time and > time again as I would imagine. If anything, this email would make > google search results that much stronger :). > > With the UA local to my network I had tested two way audio, and now > with the phone outside of network, we have no way audio. Before > discussing NAT (which is enabled on the peer), and port forwarding > (which is setup on the remote location), I would like to make sure I > fully understand all the sip.conf settings. We are using Asterisk > realtime via sip_buddies, and the fields in question are: > > (Enclosed in brackets are an example value for the setting) > > * host (dynamic): No problem here. Just wanted to mention that it's > set as such.... > * nat (yes): No problem here either.... > * defaultuser (1003 at example.com): Does the "@example.com" have to > point to the UA (i.e., (1003 at ua-public-ip), or is it just a name type > field? > * fullcontact: What to put here for a UA that is running at a remote > location with dynamic external IP? > * ipaddr (ua-public-ip): I did try setting it to the public ip of the > UA, but is that really practical? > What if I don't know where the initial registration request is coming > from? I am guessing "host=dynamic" takes care of that. > * defaultip?? > * dynamic: Should this be set to yes, or is "host=dynamic" sufficient? > > The phone registers fine, and terminates a call through our providers. > Just no audio both ways, which would suggest something more that an > RTP issue which should at least have one way outgoing audio. > > Things that have been attempted: > * Port forwarding to the phone > * Changing defaultuser to 1003 at ua-public-ip. This made our OpenSIPS > sip proxy through a fit. > > Things I will attempt today: > Calling the UA extension from an extension here > SIP trace > > Your help is greatly appreciated!!! > > Nick. >Hi Is your directmedia/canreinvite (depending on asterisk version) set to no? Regards Ish -- Ishfaq Malik <ish at pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552