switch_rtp.c - Auto Changing Port
Hi, I have a number of SIP endpoints behind NAT on a cellular network and am experiencing some odd behaviour with RTP traffic. No audio at all is present on the call for about the first 10s or so, and then the following is shown in the console;
2010-09-06 01:04:53.101801 [INFO] switch_rtp.c:2049 Auto Changing port from 10.21.6.213:4000 to 217.171.129.67:22263
After which the call works perfectly. Is there any way I can force this process to happen quicker, within the first second of the call? or is it possible to force FreeSWITCH to use the external IP from the start? Any assistance you can offer would be much appreciated. Best Regards, Ross
First, this process should happen quicker. If it doesn't, it's probably because FS does not receive any RTP traffic from your phone. Perhaps your cellular network is doing "things" in order to prevent VoIP. FS can't use the external IP from the start. FS has no way to know the exact port it should use to reach a specific device behind NAT, because until this device sends RTP, the port does not exist yet. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 06/09/2010 à 02:59, Ross McKillop a écrit :
Hi,
I have a number of SIP endpoints behind NAT on a cellular network and am experiencing some odd behaviour with RTP traffic.
No audio at all is present on the call for about the first 10s or so, and then the following is shown in the console;
2010-09-06 01:04:53.101801 [INFO] switch_rtp.c:2049 Auto Changing port from 10.21.6.213:4000 to 217.171.129.67:22263
After which the call works perfectly.
Is there any way I can force this process to happen quicker, within the first second of the call? or is it possible to force FreeSWITCH to use the external IP from the start?
Any assistance you can offer would be much appreciated.
Best Regards, Ross _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On 6 Sep 2010, at 14:33, David Ponzone wrote:
If it doesn't, it's probably because FS does not receive any RTP traffic from your phone. Perhaps your cellular network is doing "things" in order to prevent VoIP.
That's strange - because connecting to Asterisk in the same way it seems to work instantly, but Asterisk doesn't support TLS which is needed for this application. The mobile network is 3 UK who have confirmed there's nothing in place to prevent it and they've got no issue with VoIP traffic. I shall contact the vendor of the softphone on the mobile device and see if they can offer any advice. Thanks, Ross
The other fix is to tell your device to not lie about the ip/port in the SDP then it would never have to auto adj in the first place. /b On Sep 7, 2010, at 4:19 AM, Ross McKillop wrote:
On 6 Sep 2010, at 14:33, David Ponzone wrote:
If it doesn't, it's probably because FS does not receive any RTP traffic from your phone. Perhaps your cellular network is doing "things" in order to prevent VoIP.
That's strange - because connecting to Asterisk in the same way it seems to work instantly, but Asterisk doesn't support TLS which is needed for this application.
The mobile network is 3 UK who have confirmed there's nothing in place to prevent it and they've got no issue with VoIP traffic.
I shall contact the vendor of the softphone on the mobile device and see if they can offer any advice.
Thanks,
Ross
The other fix is to tell your device to not lie about the ip/port in the SDP then it would never have to auto adj in the first place. /b On Sep 7, 2010, at 4:19 AM, Ross McKillop wrote:
On 6 Sep 2010, at 14:33, David Ponzone wrote:
If it doesn't, it's probably because FS does not receive any RTP traffic from your phone. Perhaps your cellular network is doing "things" in order to prevent VoIP.
That's strange - because connecting to Asterisk in the same way it seems to work instantly, but Asterisk doesn't support TLS which is needed for this application.
The mobile network is 3 UK who have confirmed there's nothing in place to prevent it and they've got no issue with VoIP traffic.
I shall contact the vendor of the softphone on the mobile device and see if they can offer any advice.
Thanks,
Ross
Please, Brian, don't post twice the same reply! :) :) :) David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 07/09/2010 à 16:34, Brian West a écrit :
The other fix is to tell your device to not lie about the ip/port in the SDP then it would never have to auto adj in the first place.
/b
On Sep 7, 2010, at 4:19 AM, Ross McKillop wrote:
On 6 Sep 2010, at 14:33, David Ponzone wrote:
If it doesn't, it's probably because FS does not receive any RTP traffic from your phone. Perhaps your cellular network is doing "things" in order to prevent VoIP.
That's strange - because connecting to Asterisk in the same way it seems to work instantly, but Asterisk doesn't support TLS which is needed for this application.
The mobile network is 3 UK who have confirmed there's nothing in place to prevent it and they've got no issue with VoIP traffic.
I shall contact the vendor of the softphone on the mobile device and see if they can offer any advice.
Thanks,
Ross
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yes travel in time... or send your RTP packets faster. /b On Sep 5, 2010, at 8:59 PM, Ross McKillop wrote:
Is there any way I can force this process to happen quicker, within the first second of the call? or is it possible to force FreeSWITCH to use the external IP from the start?
I have same issue, with phone behind nat. The RTP arrive to FS but don't have the auto change IP/Port, so Not media can still stablish. and FS have not sent any rtp packets. In Asterisk after received 2 packets, it started to sending to rexternal IP/port . Is there any variable can setup for solve the problem? Thanks, Marhbere. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/switch-rtp-c-Auto-Changing-Por... Sent from the freeswitch-users mailing list archive at Nabble.com.
Ross, for the record, FS needs 20 packets to auto-adjust to the real source IP:port of the RTP stream. So with a typical ptime of 20ms, that's 400ms before FS learns it, so before the audio works both ways. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 06/09/2010 à 02:59, Ross McKillop a écrit :
Hi,
I have a number of SIP endpoints behind NAT on a cellular network and am experiencing some odd behaviour with RTP traffic.
No audio at all is present on the call for about the first 10s or so, and then the following is shown in the console;
2010-09-06 01:04:53.101801 [INFO] switch_rtp.c:2049 Auto Changing port from 10.21.6.213:4000 to 217.171.129.67:22263
After which the call works perfectly.
Is there any way I can force this process to happen quicker, within the first second of the call? or is it possible to force FreeSWITCH to use the external IP from the start?
Any assistance you can offer would be much appreciated.
Best Regards, Ross _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On 7 Sep 2010, at 09:13, David Ponzone wrote:
for the record, FS needs 20 packets to auto-adjust to the real source IP:port of the RTP stream. So with a typical ptime of 20ms, that's 400ms before FS learns it, so before the audio works both ways.
400ms is a lot shorter than the 10s I'm experiencing at the moment ; ) I'll try some packet captures and see what it's doing - problem is with SIP TLS it's harder to get anything meaningful out of Wireshark's SIP functions. Ross
So you should first go back to regular SIP/RTP and see if the issue is still there. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 07/09/2010 à 10:25, Ross McKillop a écrit :
On 7 Sep 2010, at 09:13, David Ponzone wrote:
for the record, FS needs 20 packets to auto-adjust to the real source IP:port of the RTP stream. So with a typical ptime of 20ms, that's 400ms before FS learns it, so before the audio works both ways.
400ms is a lot shorter than the 10s I'm experiencing at the moment ; )
I'll try some packet captures and see what it's doing - problem is with SIP TLS it's harder to get anything meaningful out of Wireshark's SIP functions.
Ross
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
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Brian West
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David Ponzone
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MarhBere
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Ross McKillop