No audio after Remote SDP:
Hi! I'm sure this is a NAT issue, but I'm not sure what options to use. I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102. However, after FS receives the Remote SDP the audio stops (ring tone stops in my case) 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel sofia/internal/sip:2001@192.168.1.3:56885 entering state [completing][200] 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: v=0 o=- 18490612 18490612 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16432 RTP/AVP 2 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 I notice that the ip address in the o and c fields indicate a local IP address. Should this IP address be an external IP address of the 3102 instead? Thanks
On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote:
DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a grain of salt.
Welcome to the community.
I have a similar setup (and problem) - the wiki documentation refers to it as "double nat". Like you, my FS and client are behind different NATs and I can register my remote endpoint and make calls (in my case, to the the FS demo ivr at 5000).
Since your external endpoint (spa3102) is registering, you've likely setup your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc).
Your endpoint need only insert rport and FreeSWITCH will do the right thing.
1) Setup stun on your remote endpoint (spa3102 in your case) 2) Add <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> to the directory xml file that describes your spa3102 endpoint
The device supports STUN also its highly recommended your device know how to overcome its own NAT. I personally do not believe its the registrars place to overcome an endpoints nat... puts undue burden on the registar.
Option 1 worked for me right away (eyebeam in my case) and, as expected, the remote sdp had the correct (remote) IP address, since the endpoint is using stun to correctly identify its IP address to FS. However, option 2 has not made a difference (for me). Is it just me or is it strange that SIP works without stun, but RTP doesn't?
I guess I've been spoiled by the way Asterisk handles NAT and was hopeful that NDLB-connectile-dysfunction would behave similarly, so I wouldn't have to tell users to setup stun on their clients. Maybe a FS user with some experience with this type of NAT setup and these settings can help. I'd be interested in knowing how to correctly setup remote NATted endpoints without stun - or, at least, hear from someone that this setting works for them without stun.
Anyway, hope this helps you with your SPA3102.
Bottom line is enable rport and use stun on the SPA and it'll just work. /b
Thanks Brian and Gad, I have stun set and if I do a 'sofia status profile internal', I see the external IP address of the 3102 ATA, so I assume that stun is working correctly on the SPA3102. These are the options that I have set (according to the 3102 manual). • Handle VIA received: yes • Handle VIA rport: yes • Insert VIA received: yes • Insert VIA rport: yes • Substitute VIA Addr: yes • Send Resp To Src Port: yes • STUN Enable: Choose yes. • STUN Server: stun.freeswitch.org I assume that is all is needed? On Mon, Dec 21, 2009 at 9:36 AM, Brian West <brian@freeswitch.org> wrote:
On Dec 20, 2009, at 2:53 PM, Gad Bentolila wrote:
DISCLAIMER: I'm REALLY new to FreeSwitch, so please take my advice with a grain of salt.
Welcome to the community.
I have a similar setup (and problem) - the wiki documentation refers to it as "double nat". Like you, my FS and client are behind different NATs and I can register my remote endpoint and make calls (in my case, to the the FS demo ivr at 5000).
Since your external endpoint (spa3102) is registering, you've likely setup your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc).
Your endpoint need only insert rport and FreeSWITCH will do the right thing.
1) Setup stun on your remote endpoint (spa3102 in your case) 2) Add <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> to the directory xml file that describes your spa3102 endpoint
The device supports STUN also its highly recommended your device know how to overcome its own NAT. I personally do not believe its the registrars place to overcome an endpoints nat... puts undue burden on the registar.
Option 1 worked for me right away (eyebeam in my case) and, as expected, the remote sdp had the correct (remote) IP address, since the endpoint is using stun to correctly identify its IP address to FS. However, option 2 has not made a difference (for me). Is it just me or is it strange that SIP works without stun, but RTP doesn't?
I guess I've been spoiled by the way Asterisk handles NAT and was hopeful that NDLB-connectile-dysfunction would behave similarly, so I wouldn't have to tell users to setup stun on their clients. Maybe a FS user with some experience with this type of NAT setup and these settings can help. I'd be interested in knowing how to correctly setup remote NATted endpoints without stun - or, at least, hear from someone that this setting works for them without stun.
Anyway, hope this helps you with your SPA3102.
Bottom line is enable rport and use stun on the SPA and it'll just work. /b
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Can you get me siptraces please. /b On Dec 20, 2009, at 5:54 PM, Mark Campbell-Smith wrote:
Thanks Brian and Gad,
I have stun set and if I do a 'sofia status profile internal', I see the external IP address of the 3102 ATA, so I assume that stun is working correctly on the SPA3102.
These are the options that I have set (according to the 3102 manual).
• Handle VIA received: yes • Handle VIA rport: yes • Insert VIA received: yes • Insert VIA rport: yes • Substitute VIA Addr: yes • Send Resp To Src Port: yes • STUN Enable: Choose yes. • STUN Server: stun.freeswitch.org
I assume that is all is needed?
Sorry for the delay. Christmas stuff! I have put the log at http://pastebin.freeswitch.org/11632 The SPA3102 has remote IP address 11.11.11.11 (in the log), and internal ip address 192.168.1.3 My local ATA has external IP address 124.11.11.11 (in the log) and internal ip address 192.168.1.120 I have all these set:
• Handle VIA received: yes • Handle VIA rport: yes • Insert VIA received: yes • Insert VIA rport: yes • Substitute VIA Addr: yes • Send Resp To Src Port: yes • STUN Enable: Choose yes. • STUN Server: stun.freeswitch.org
When I set 'Nat Mapping Enable' under tab Line 1 in the SPA3102, I get the following trace. This is all I see and then registration fails. freeswitch@internal> recv 545 bytes from tls/[11.11.11.11]:56886 at 11:12:22.924395: ------------------------------------------------------------------------ REGISTER sip:myddns.dydns.org:442 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.3:442;branch=z9hG4bK-4c71dcba;rport From: 2001 <sip:2001@myddns.dydns.org:442>;tag=2f998bc591c1321o0 To: 2001 <sip:2001@myddns.dydns.org:442> Call-ID: 115e0ffa-a538d31b@192.168.1.3 CSeq: 35409 REGISTER Max-Forwards: 70 Contact: 2001 <sip:2001@192.168.1.3:442;transport=tls>;expires=600 User-Agent: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces ------------------------------------------------------------------------ send 678 bytes to tls/[11.11.11.11]:56886 at 11:12:22.972012: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.1.3:442;branch=z9hG4bK-4c71dcba;rport=56886;received=11.11.11.11 From: 2001 <sip:2001@myddns.dydns.org:442>;tag=2f998bc591c1321o0 To: 2001 <sip:2001@myddns.dydns.org:442>;tag=07c0ymHUv7KXH Call-ID: 115e0ffa-a538d31b@192.168.1.3 CSeq: 35409 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15490 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="myddns.dydns.org", nonce="35a9849e-f07d-11de-88a5-dbc3ffce4ce8", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ ------------------------------------------------------------------------ Thanks! On Tue, Dec 22, 2009 at 2:03 AM, Brian West <brian@freeswitch.org> wrote:
Can you get me siptraces please. /b On Dec 20, 2009, at 5:54 PM, Mark Campbell-Smith wrote:
Thanks Brian and Gad,
I have stun set and if I do a 'sofia status profile internal', I see the external IP address of the 3102 ATA, so I assume that stun is working correctly on the SPA3102.
These are the options that I have set (according to the 3102 manual).
• Handle VIA received: yes • Handle VIA rport: yes • Insert VIA received: yes • Insert VIA rport: yes • Substitute VIA Addr: yes • Send Resp To Src Port: yes • STUN Enable: Choose yes. • STUN Server: stun.freeswitch.org
I assume that is all is needed?
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If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. I would have to see what happens after the 401 to see if it really did fail. /b On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote:
This is all I see and then registration fails.
Hi Brian, But that is all I see... I don't see any further messages after the 401 (when I have 'NAT Mapping Enabled' selected) ..... I realise that this is part of the authentication procedure, but it stops at the 401/Authorized message What about in the pastebin? Do you see anything there? This is when the call gets setup, but the IP address in the SDP says a private IP address http://pastebin.freeswitch.org/11632 On Tue, Dec 29, 2009 at 2:25 AM, Brian West <brian@freeswitch.org> wrote:
If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. I would have to see what happens after the 401 to see if it really did fail.
/b
On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote:
This is all I see and then registration fails.
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are you using more than one profile here? if so can you repeat the trace with siptrace on in both profiles. I notice this on the trace: 1. Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-ee832ee7 2. From: 1000 <sip:1000@192.168.1.120>;tag=70c4b62bd9443c9fo0 and this: sofia_glue.c:2344 AUDIO RTP [sofia/internal/1000@192.168.1.120] 192.168.1 .120 port 28490 -> 192.168.1.121 port 16464 codec: 2 ms: 20 Are you behind double nat and or is FS also behind nat? To address the gentleman who mentioned he was spoiled by asterisk. Keep in mind we have several advanced nat techniques but many of them break other situations and they are not all turned on at once with one parameter like in asterisk, we prefer you use the appropriate ones based on the situation. On Wed, Dec 30, 2009 at 3:42 AM, Mark Campbell-Smith < mcampbellsmith@gmail.com> wrote:
Hi Brian,
But that is all I see... I don't see any further messages after the 401 (when I have 'NAT Mapping Enabled' selected) ..... I realise that this is part of the authentication procedure, but it stops at the 401/Authorized message
What about in the pastebin? Do you see anything there? This is when the call gets setup, but the IP address in the SDP says a private IP address http://pastebin.freeswitch.org/11632
On Tue, Dec 29, 2009 at 2:25 AM, Brian West <brian@freeswitch.org> wrote:
If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. I would have to see what happens after the 401 to see if it really did fail.
/b
On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote:
This is all I see and then registration fails.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Hi Anthony, The only profiles I have defined are external and internal. These should be using internal... 192.168.1.120 is the FS box, which is NAT'd. Never had any problems with this being NAT'd though 192.168.1.121 is a PAP2 ATA connected to FS I don't use proxy media. I am trying to call an SPA3102, which is on the internet and NAT'd (external IP address 11.11.11.11 in the trace and internal/private ip address of 192.168.1.3). Thanks! On Thu, Dec 31, 2009 at 1:04 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
are you using more than one profile here? if so can you repeat the trace with siptrace on in both profiles.
I notice this on the trace:
Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-ee832ee7 From: 1000 <sip:1000@192.168.1.120>;tag=70c4b62bd9443c9fo0
and this:
sofia_glue.c:2344 AUDIO RTP [sofia/internal/1000@192.168.1.120] 192.168.1.120 port 28490 -> 192.168.1.121 port 16464 codec: 2 ms: 20
Are you behind double nat and or is FS also behind nat?
To address the gentleman who mentioned he was spoiled by asterisk. Keep in mind we have several advanced nat techniques but many of them break other situations and they are not all turned on at once with one parameter like in asterisk, we prefer you use the appropriate ones based on the situation.
On Wed, Dec 30, 2009 at 3:42 AM, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Hi Brian,
But that is all I see... I don't see any further messages after the 401 (when I have 'NAT Mapping Enabled' selected) ..... I realise that this is part of the authentication procedure, but it stops at the 401/Authorized message
What about in the pastebin? Do you see anything there? This is when the call gets setup, but the IP address in the SDP says a private IP address http://pastebin.freeswitch.org/11632
On Tue, Dec 29, 2009 at 2:25 AM, Brian West <brian@freeswitch.org> wrote:
If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. I would have to see what happens after the 401 to see if it really did fail.
/b
On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote:
This is all I see and then registration fails.
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please. /b On Dec 30, 2009, at 2:13 PM, Mark Campbell-Smith wrote:
Hi Anthony,
The only profiles I have defined are external and internal. These should be using internal...
192.168.1.120 is the FS box, which is NAT'd. Never had any problems with this being NAT'd though 192.168.1.121 is a PAP2 ATA connected to FS
I don't use proxy media.
I am trying to call an SPA3102, which is on the internet and NAT'd (external IP address 11.11.11.11 in the trace and internal/private ip address of 192.168.1.3).
Thanks!
Hi! Both are auto-nat: <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> FreeSWITCH Version 1.0.trunk (15490) However, isn't it the IP address that is reported by the remote SPA3102 that is incorrect? Or? On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian@freeswitch.org> wrote:
show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please.
/b
On Dec 30, 2009, at 2:13 PM, Mark Campbell-Smith wrote:
Hi Anthony,
The only profiles I have defined are external and internal. These should be using internal...
192.168.1.120 is the FS box, which is NAT'd. Never had any problems with this being NAT'd though 192.168.1.121 is a PAP2 ATA connected to FS
I don't use proxy media.
I am trying to call an SPA3102, which is on the internet and NAT'd (external IP address 11.11.11.11 in the trace and internal/private ip address of 192.168.1.3).
Thanks!
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Are you behind a nat-pmp/upnp router? /b On Jan 2, 2010, at 1:30 AM, Mark Campbell-Smith wrote:
Hi!
Both are auto-nat: <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/>
FreeSWITCH Version 1.0.trunk (15490)
However, isn't it the IP address that is reported by the remote SPA3102 that is incorrect? Or?
On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian@freeswitch.org> wrote:
show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please.
/b
I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102. However, after FS receives the Remote SDP the audio stops (ring tone stops in my case) The FS router is upnp enabled. The SPA3102 router is NOT upnp enabled (SPA3102 does not support upnp anyway I think). On Sun, Jan 3, 2010 at 3:03 AM, Brian West <brian@freeswitch.org> wrote:
Are you behind a nat-pmp/upnp router? /b On Jan 2, 2010, at 1:30 AM, Mark Campbell-Smith wrote:
Hi!
Both are auto-nat: <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/>
FreeSWITCH Version 1.0.trunk (15490)
However, isn't it the IP address that is reported by the remote SPA3102 that is incorrect? Or?
On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian@freeswitch.org> wrote:
show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please.
/b
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OK.. I have looked at this some more... Below is the syslog from the SPA3102: THIS WORKS Jan 5 21:18:17 92.xx.xx.xx [0]->124.xxx.xxx.xxx:442(544) Jan 5 21:18:17 92.xx.xx.xx REGISTER sip:myddns.dydns.org:442 SIP/2.0 Via: SIP/2.0/TLS 92.xx.xx.xx:5069;branch=z9hG4bK-7b7ebbf9 From: 2001 <sip:2001@myddns.dydns.org:442>;tag=b69c38c549e24c42o0 To: 2001 <sip:2001@myddns.dydns.org:442> Call-ID: e6b918dc-71d58fe3@192.168.1.3 CSeq: 44358 REGISTER Max-Forwards: 70 Contact: 2001 <sip:2001@92.xx.xx.xx:5069;transport=tls>;expires=600 User-Agent: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Jan 5 21:18:17 92.xx.xx.xx Jan 5 21:18:17 92.xx.xx.xx Jan 5 21:18:17 92.xx.xx.xx [0]<<124.xxx.xxx.xxx:442(658) Jan 5 21:18:17 92.xx.xx.xx [0]<<124.xxx.xxx.xxx:442(658) Jan 5 21:18:17 92.xx.xx.xx SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 92.xx.xx.xx:5069;branch=z9hG4bK-7b7ebbf9;rport=5069 From: 2001 <sip:2001@myddns.dydns.org:442>;tag=b69c38c549e24c42o0 To: 2001 <sip:2001@myddns.dydns.org:442>;tag=Bav1HeBr3jm3B Call-ID: e6b918dc-71d58fe3@192.168.1.3 CSeq: 44358 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16131 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="myddns.dydns.org", nonce="a4128380-f9e3-11de-99eb-53ce5686ac9a", algorithm=MD5, qop="auth" Content-Length: 0 Jan 5 21:18:17 92.xx.xx.xx Jan 5 21:18:17 92.xx.xx.xx Jan 5 21:18:17 92.xx.xx.xx [0]->124.xxx.xxx.xxx:442(782) Jan 5 21:18:17 92.xx.xx.xx [0]->124.xxx.xxx.xxx:442(782) Jan 5 21:18:17 92.xx.xx.xx REGISTER sip:myddns.dydns.org:442 SIP/2.0 Via: SIP/2.0/TLS 92.xx.xx.xx:5069;branch=z9hG4bK-18f00822 From: 2001 <sip:2001@myddns.dydns.org:442>;tag=b69c38c549e24c42o0 To: 2001 <sip:2001@myddns.dydns.org:442> Call-ID: e6b918dc-71d58fe3@192.168.1.3 CSeq: 44359 REGISTER Max-Forwards: 70 Authorization: Digest username="2001",realm="myddns.dydns.org",nonce="a4128380-f9e3-11de-99eb-53ce5686ac9a",uri="sip:myddns.dydns.org:442",algorithm=MD5,response="324ee93184ae202be4a209f5a9255229",qop=auth,nc=00000001,cnonce="804844b" Contact: 2001 <sip:2001@92.xx.xx.xx:5069;transport=tls>;expires=600 User-Agent: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces THIS DOES NOT WORK Jan 5 22:05:48 92.xxx.xxx.xxx REGISTER sip:myddns.dydns.org:442 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.3:5070;branch=z9hG4bK-faf8477a From: 2001 <sip:2001@myddns.dydns.org:442>;tag=b065057e3ed0befdo0 To: 2001 <sip:2001@myddns.dydns.org:442> Call-ID: e6b918dc-71d58fe3@192.168.1.3 CSeq: 44435 REGISTER Max-Forwards: 70 Contact: 2001 <sip:2001@192.168.1.3:5070;transport=tls>;expires=600 User-Agent: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Jan 5 22:05:48 92.xxx.xxx.xxx SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.1.3:5070;branch=z9hG4bK-faf8477a;received=92.xxx.xxx.xxx;rport=5070 From: 2001 <sip:2001@myddns.dydns.org:442>;tag=b065057e3ed0befdo0 To: 2001 <sip:2001@myddns.dydns.org:442>;tag=N51jvyeca9Umj Call-ID: e6b918dc-71d58fe3@192.168.1.3 CSeq: 44435 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16131 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="myddns.dydns.org", nonce="476097b0-f9ea-11de-99fd-53ce5686ac9a", algorithm=MD5, qop="auth" Content-Length: 0 Jan 5 22:05:48 92.xxx.xxx.xxx Jan 5 22:05:48 92.xxx.xxx.xxx Jan 5 22:05:58 92.xxx.xxx.xxx [0]->124.xxx.xxx.xxx:442(783) Jan 5 22:05:58 92.xxx.xxx.xxx [0]->124.xxx.xxx.xxx:442(783) Jan 5 22:05:58 92.xxx.xxx.xxx REGISTER sip:myddns.dydns.org:442 SIP/2.0 Via: SIP/2.0/TLS 92.xxx.xxx.xxx:5070;branch=z9hG4bK-bfc992b3 From: 2001 <sip:2001@myddns.dydns.org:442>;tag=b065057e3ed0befdo0 To: 2001 <sip:2001@myddns.dydns.org:442> Call-ID: e6b918dc-71d58fe3@192.168.1.3 CSeq: 44436 REGISTER Max-Forwards: 70 Authorization: Digest username="2001",realm="myddns.dydns.org",nonce="476097b0-f9ea-11de-99fd-53ce5686ac9a",uri="sip:myddns.dydns.org:442",algorithm=MD5,response="466ea78ac2ccddca991a5c3d4d021bed",qop=auth,nc=00000001,cnonce="ebd80711" Contact: 2001 <sip:2001@92.xxx.xxx.xxx:5070;transport=tls>;expires=600 User-Agent: Linksys/SPA3102-5.1.10(GW) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces The difference was a hard coded external IP address in the first session that worked. However I can't have it set like this as the IP address is not static. The second REGISTER in the Not Working session seems to be ignored by FreeSwitch (I don't see it in the logs of FS either). Is there something in this Register that causes FS to ignore it? Thanks On Sun, Jan 3, 2010 at 10:19 AM, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102.
However, after FS receives the Remote SDP the audio stops (ring tone stops in my case)
The FS router is upnp enabled. The SPA3102 router is NOT upnp enabled (SPA3102 does not support upnp anyway I think).
On Sun, Jan 3, 2010 at 3:03 AM, Brian West <brian@freeswitch.org> wrote:
Are you behind a nat-pmp/upnp router? /b On Jan 2, 2010, at 1:30 AM, Mark Campbell-Smith wrote:
Hi!
Both are auto-nat: <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/>
FreeSWITCH Version 1.0.trunk (15490)
However, isn't it the IP address that is reported by the remote SPA3102 that is incorrect? Or?
On Thu, Dec 31, 2009 at 7:21 AM, Brian West <brian@freeswitch.org> wrote:
show me the ext-rtp-ip and ext-sip-ip settings you're using along with SVN rev please.
/b
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You'll need to fix your device to know its IP and it should stop doing that. /b On Dec 20, 2009, at 5:58 AM, Mark Campbell-Smith wrote:
Hi!
I'm sure this is a NAT issue, but I'm not sure what options to use.
I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102.
However, after FS receives the Remote SDP the audio stops (ring tone stops in my case)
2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel sofia/internal/sip:2001@192.168.1.3:56885 entering state [completing][200] 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: v=0 o=- 18490612 18490612 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16432 RTP/AVP 2 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20
I notice that the ip address in the o and c fields indicate a local IP address. Should this IP address be an external IP address of the 3102 instead?
Thanks
participants (4)
-
Anthony Minessale
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Brian West
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Gad Bentolila
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Mark Campbell-Smith