No Media in SIP Calling from 3G Mobile Internet to Freeswitch
Hi, Currently, I face an issue that I could not hear the audio sound at all when I try to call to FreeSWITCH (9386) via its public domain name from the SIP client of my mobile phone with mobile internet. In this case, SIP registration with TLS and media encryption with SRTP is working perfectly because I could see the secure symbol in Linphone *but* I could not hear any audio at all. After 30 seconds of the silent call, it hanged up automatically. *SIP Client on the 3G Mobile Internet > Freeswitch (tls:voip.test.com <http://voip.test.com>, SRTP) * *SIP Registration Sucess, Call No Media* My question is that is this the problem of NAT? If so, it would be great if you could help me solve this problem. Any help would be really appreciated. Best regards, Chhorm Chhatra
Hi, Yes, that sounds like a network (NAT/Firewall) issue. Your mobile provider may use a chain of several NATs and not every one of them is working properly. Try VPN or another SIP client. Regards, Alex Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303 Email: alex@freeswitch.com Website: https://www.signalwire.com On Sat, Feb 2, 2019 at 2:44 AM Chhorm Chhatra <ch.chhatra@gmail.com> wrote:
Hi, Currently, I face an issue that I could not hear the audio sound at all when I try to call to FreeSWITCH (9386) via its public domain name from the SIP client of my mobile phone with mobile internet. In this case, SIP registration with TLS and media encryption with SRTP is working perfectly because I could see the secure symbol in Linphone *but* I could not hear any audio at all. After 30 seconds of the silent call, it hanged up automatically.
*SIP Client on the 3G Mobile Internet > Freeswitch (tls:voip.test.com <http://voip.test.com>, SRTP) * *SIP Registration Sucess, Call No Media*
My question is that is this the problem of NAT? If so, it would be great if you could help me solve this problem. Any help would be really appreciated.
Best regards, Chhorm Chhatra _________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
Adding a stun can help too, if it is Linphone on Android then please add stun.linphone.org in stun settings further disable AVFP & ICE On Sat, Feb 2, 2019, 9:32 AM Alexey Sibyakin <alex@freeswitch.com wrote:
Hi,
Yes, that sounds like a network (NAT/Firewall) issue. Your mobile provider may use a chain of several NATs and not every one of them is working properly. Try VPN or another SIP client.
Regards,
Alex
Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303 Email: alex@freeswitch.com Website: https://www.signalwire.com
On Sat, Feb 2, 2019 at 2:44 AM Chhorm Chhatra <ch.chhatra@gmail.com> wrote:
Hi, Currently, I face an issue that I could not hear the audio sound at all when I try to call to FreeSWITCH (9386) via its public domain name from the SIP client of my mobile phone with mobile internet. In this case, SIP registration with TLS and media encryption with SRTP is working perfectly because I could see the secure symbol in Linphone *but* I could not hear any audio at all. After 30 seconds of the silent call, it hanged up automatically.
*SIP Client on the 3G Mobile Internet > Freeswitch (tls:voip.test.com <http://voip.test.com>, SRTP) * *SIP Registration Sucess, Call No Media*
My question is that is this the problem of NAT? If so, it would be great if you could help me solve this problem. Any help would be really appreciated.
Best regards, Chhorm Chhatra _________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
Dear Alex and Faisal, Thank you so much for your suggested solutions. In my case, there were some issues with firewall rules. I had to allow the UDP and TCP ports for SIP and RTP signals. For more information to those who might face the same problem, please see the attached image. Best regards, Chhorm Chhatra On Sun, 3 Feb 2019 at 04:27, Faisal Hanif <imfanee@gmail.com> wrote:
Adding a stun can help too, if it is Linphone on Android then please add stun.linphone.org in stun settings further disable AVFP & ICE
On Sat, Feb 2, 2019, 9:32 AM Alexey Sibyakin <alex@freeswitch.com wrote:
Hi,
Yes, that sounds like a network (NAT/Firewall) issue. Your mobile provider may use a chain of several NATs and not every one of them is working properly. Try VPN or another SIP client.
Regards,
Alex
Alex Sibyakin | Support Engineer SignalWire | 228 Hamilton Ave 3rd Floor, Palo Alto, CA 94303 Email: alex@freeswitch.com Website: https://www.signalwire.com
On Sat, Feb 2, 2019 at 2:44 AM Chhorm Chhatra <ch.chhatra@gmail.com> wrote:
Hi, Currently, I face an issue that I could not hear the audio sound at all when I try to call to FreeSWITCH (9386) via its public domain name from the SIP client of my mobile phone with mobile internet. In this case, SIP registration with TLS and media encryption with SRTP is working perfectly because I could see the secure symbol in Linphone *but* I could not hear any audio at all. After 30 seconds of the silent call, it hanged up automatically.
*SIP Client on the 3G Mobile Internet > Freeswitch (tls:voip.test.com <http://voip.test.com>, SRTP) * *SIP Registration Sucess, Call No Media*
My question is that is this the problem of NAT? If so, it would be great if you could help me solve this problem. Any help would be really appreciated.
Best regards, Chhorm Chhatra _________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
_________________________________________________________________________ Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
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 https://freeswitch.com
participants (3)
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Alexey Sibyakin
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Chhorm Chhatra
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Faisal Hanif