dslepnev Posted February 17, 2009 Report Share Posted February 17, 2009 Hi, We trying to configure pbxnsip to work with remote users who would like to work at home. Most of remote users have internet connection behind NAT. It causes problem when users don't hear audio at all. What happens: This is INVITE from remote user. This INVITE coming from real address. 1 0.000000 195.146.74.104 192.168.22.43 SIP/SDP Request: INVITE sip:112@tw;user=phone, with session description pbxnsip answering to this address. Good. 2 0.003732 192.168.22.43 195.146.74.104 SIP Status: 100 Trying 3 0.010382 192.168.22.43 195.146.74.104 SIP/SDP Status: 200 Ok, with session description and trying to send RTP to private address. This address taken by pbxnsip from INVITE. 4 0.021485 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24774, Time=3241200, Mark 6 0.041015 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24775, Time=3241360 Sure that pbxnsip working fully according SIP RFC - sends RTP to IP address from INVITE/SDP. Quesion is: maybe possible to configure pbxnsip to send RTP to addres from INVITE received? As part of Mini-SBC function.. Quote Link to comment Share on other sites More sharing options...
YMSL Posted February 17, 2009 Report Share Posted February 17, 2009 Hi, We trying to configure pbxnsip to work with remote users who would like to work at home. Most of remote users have internet connection behind NAT. It causes problem when users don't hear audio at all. What happens: This is INVITE from remote user. This INVITE coming from real address. 1 0.000000 195.146.74.104 192.168.22.43 SIP/SDP Request: INVITE sip:112@tw;user=phone, with session description pbxnsip answering to this address. Good. 2 0.003732 192.168.22.43 195.146.74.104 SIP Status: 100 Trying 3 0.010382 192.168.22.43 195.146.74.104 SIP/SDP Status: 200 Ok, with session description and trying to send RTP to private address. This address taken by pbxnsip from INVITE. 4 0.021485 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24774, Time=3241200, Mark 6 0.041015 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24775, Time=3241360 Sure that pbxnsip working fully according SIP RFC - sends RTP to IP address from INVITE/SDP. Quesion is: maybe possible to configure pbxnsip to send RTP to addres from INVITE received? As part of Mini-SBC function.. I have faced the same situation, to resolve I didi the following: 1- use a router that support DMZ and set to the adress of your PBX server - work fine OR 2- set your router to allow ports for your PBX - all of them RTP is transporting audio you can also set your PBX to use the following route table : under Admin>Ports set "SIP IP Replacement List" to first IP of your PBX and IP of your router - something like 192.168.0.1/xxx.xxx.xxx.xxx -> PBXIP/RouterIP restart PBX and It should work fine This run fine on my side BTW you should have a very very very good Internet connection if you want to keep audio quality for all external registration on your PBX otherwise it will be more like a cell phone quality, forcing codec might help. hope it help! Quote Link to comment Share on other sites More sharing options...
pbx support Posted February 17, 2009 Report Share Posted February 17, 2009 Hi, We trying to configure pbxnsip to work with remote users who would like to work at home. Most of remote users have internet connection behind NAT. It causes problem when users don't hear audio at all. What happens: This is INVITE from remote user. This INVITE coming from real address. 1 0.000000 195.146.74.104 192.168.22.43 SIP/SDP Request: INVITE sip:112@tw;user=phone, with session description pbxnsip answering to this address. Good. 2 0.003732 192.168.22.43 195.146.74.104 SIP Status: 100 Trying 3 0.010382 192.168.22.43 195.146.74.104 SIP/SDP Status: 200 Ok, with session description and trying to send RTP to private address. This address taken by pbxnsip from INVITE. 4 0.021485 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24774, Time=3241200, Mark 6 0.041015 192.168.22.43 192.168.1.25 RTP Payload type=ITU-T G.711 PCMU, SSRC=1237099670, Seq=24775, Time=3241360 Sure that pbxnsip working fully according SIP RFC - sends RTP to IP address from INVITE/SDP. Quesion is: maybe possible to configure pbxnsip to send RTP to addres from INVITE received? As part of Mini-SBC function.. Based on the above log, it looks like SIP messages are not reaching the phones. I do not see an ACK from the phone for the 200 OK. Please make sure that the home router is not blocking the traffic from PBX. Generally, SIP/RTP uses UDP. So you have to make sure that port UDP 5060 is not blocked and for RTP, 49152-64512 are not blocked. Please refer Admin->settings->ports page to see these values. Quote Link to comment Share on other sites More sharing options...
dslepnev Posted February 17, 2009 Author Report Share Posted February 17, 2009 Based on the above log, it looks like SIP messages are not reaching the phones. I do not see an ACK from the phone for the 200 OK. Please make sure that the home router is not blocking the traffic from PBX. Generally, SIP/RTP uses UDP. So you have to make sure that port UDP 5060 is not blocked and for RTP, 49152-64512 are not blocked. Please refer Admin->settings->ports page to see these values. SIP messages reaching ip phones. I checked that home router do not blocking the traffic from pbx. Pbxnsip send media traffic to internal IP address, and this media can't be delivered to the ip phone. This is the problem. Sorry, I can't attach the trace "Upload failed. You are not permitted to upload this type of file". Quote Link to comment Share on other sites More sharing options...
dslepnev Posted February 17, 2009 Author Report Share Posted February 17, 2009 I have faced the same situation, to resolve I didi the following: 1- use a router that support DMZ and set to the adress of your PBX server - work fine I need to set routers which supports DMZ on all user's locations. Not best way. SIP ALG? OR 2- set your router to allow ports for your PBX - all of them RTP is transporting audio There is problem not in ports. Problem in addresses, which coming inside SIP messages to the PBX. you can also set your PBX to use the following route table : under Admin>Ports set "SIP IP Replacement List" to first IP of your PBX and IP of your router - something like 192.168.0.1/xxx.xxx.xxx.xxx -> PBXIP/RouterIP restart PBX and It should work fine 50/50. It can help in case when static IP used. Most of ISP's changes external IP addresses of home users (DSL) once a day. I need to seek this addresses.... This run fine on my side BTW you should have a very very very good Internet connection if you want to keep audio quality for all external registration on your PBX otherwise it will be more like a cell phone quality, forcing codec might help. G.711 used only for tests. Sure we will use G.729 in future. hope it help! Quote Link to comment Share on other sites More sharing options...
YMSL Posted February 17, 2009 Report Share Posted February 17, 2009 I need to set routers which supports DMZ on all user's locations. Not best way. SIP ALG? No just on the PBX side, on the client side it should be working with no change at all, If needed only you just have to route phone IP with 5060-5061 and RDT ports. better to use static IP for the phone! 50/50. It can help in case when static IP used. Most of ISP's changes external IP addresses of home users (DSL) once a day. I need to seek this addresses.... I use no-ip.com dynamic DNS it prevent loosing adresse everyday and I use PBX IP/DNS name Quote Link to comment Share on other sites More sharing options...
pbx support Posted February 17, 2009 Report Share Posted February 17, 2009 SIP messages reaching ip phones. I checked that home router do not blocking the traffic from pbx. Pbxnsip send media traffic to internal IP address, and this media can't be delivered to the ip phone. This is the problem. Sorry, I can't attach the trace "Upload failed. You are not permitted to upload this type of file". You can send the trace to support@pbxnsip.com Quote Link to comment Share on other sites More sharing options...
dslepnev Posted February 17, 2009 Author Report Share Posted February 17, 2009 You can send the trace to support@pbxnsip.com I sent the trace already to support@pbxnsip.com Quote Link to comment Share on other sites More sharing options...
pbx support Posted February 17, 2009 Report Share Posted February 17, 2009 I sent the trace already to support@pbxnsip.com That trace does not contain ACK for the 200 OK message. That's why I think that the phone did not get any message from the PBX. Since you are using snom phone here, you can look at the SIP trace and make sure that the phone is getting messages from the PBX. You can also run the pcap trace on the snom phone (debugging purposes only) Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 17, 2009 Report Share Posted February 17, 2009 http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses Quote Link to comment Share on other sites More sharing options...
Porter Posted January 20, 2011 Report Share Posted January 20, 2011 http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses Is there a current valid URL for this resource? I am seeing this link in many places, and can't find the new location of the document. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 20, 2011 Report Share Posted January 20, 2011 Try http://kiwi.pbxnsip.com/index.php/Office_with_private_and_public_IP_addresses, we need to move this one also to the new Wiki. Quote Link to comment Share on other sites More sharing options...
Porter Posted January 21, 2011 Report Share Posted January 21, 2011 Try http://kiwi.pbxnsip.com/index.php/Office_with_private_and_public_IP_addresses, we need to move this one also to the new Wiki. Thanks for the link! Coincidentally, I found it myself when searching around earlier, it took some digging. I came back to post the link and found you'd beaten me to it... but thanks anyway! Quote Link to comment Share on other sites More sharing options...
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