Jump to content

I broke TLS


Steve B
 Share

Recommended Posts

Hi guys,

 

I have been having a bit of an issue when it comes to one way audio through an incoming vitelity trunk on a TLS connected phone.

 

So I was playing around with the ports and pow, no audio at all in TLS, I can get two way audio in UDP. And when I say no audio, no audio on the trunk or when dialing voicemail. If the call makes it to my cellphone, no problem. I have factory reset the PBX and no luck with audio in TLS, am I going to have to wipe the system and start again? Also, no audio from webRTC now either.

 

I have changed routers to make sure this wasn't the issue. Still could be but UDP calls to and from the Vodia PBX and another PBX work flawlessly.

 

The router's nat profile is port-restricted cone on both routers. I put the phone in the DMZ and no help there.

 

Thanks,

Steve

Link to comment
Share on other sites

Hi, Steve can you check this article and run the PCAP setting on the extension level when you using TLS

 

http://wiki.snomone.com/index.php?title=Extension_PCAP

 

You can also activate the PCAP setting on the trunk level for inbound and outbound call.

 

http://wiki.snomone.com/index.php?title=Trunk_PCAP_Generation

 

Best regards

Link to comment
Share on other sites

Hi, Steve can you check this article and run the PCAP setting on the extension level when you using TLS

 

http://wiki.snomone.com/index.php?title=Extension_PCAP

 

You can also activate the PCAP setting on the trunk level for inbound and outbound call.

 

http://wiki.snomone.com/index.php?title=Trunk_PCAP_Generation

 

Best regards

 

I sent you a PM with the PCAP.

Link to comment
Share on other sites

Looks like the phone is remote? The phone send it's SDP with 192.168.1.182 and the PBX is sending it's SDP with 24.119.220.155. The PBX cannot send RTP traffic to a local IP.


Check if you have a firewall set at the server level and on the router you can turn them off for testing services and try your the VM test.

i this case if the phone was remote it would be sending SDP based on its Dynamic IP.



INVITE sip:*97@voip.barrettsys.com;user=phone SIP/2.0


Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport


From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz


To: <sip:*97@voip.barrettsys.com;user=phone>


Call-ID: 52b0f3c5d925-dsalfsaxbvip


CSeq: 1 INVITE


Max-Forwards: 70


Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1


X-Serialnumber: 000413244C69


P-Key-Flags: keys="3"


User-Agent: snom320/8.7.3.25


Accept: application/sdp


Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE


Allow-Events: talk, hold, refer, call-info


Supported: timer, 100rel, replaces, from-change


Session-Expires: 3600;refresher=uas


Min-SE: 90


Proxy-Require: buttons-snom320


Content-Type: application/sdp


Content-Length: 502




v=0


o=root 584979493 584979493 IN IP4 192.168.1.182


s=call


c=IN IP4 192.168.1.182


t=0 0


m=audio 50780 RTP/AVP 0 8 3 9 99 18 101


a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6Dwq4Jkn2h+aiF4YkKmZnmu9wwiBVgl01psqpScw


a=rtpmap:0 PCMU/8000


a=rtpmap:8 PCMA/8000


a=rtpmap:3 GSM/8000


a=rtpmap:9 G722/8000


a=rtpmap:99 G726-32/8000


a=rtpmap:18 G729/8000


a=fmtp:18 annexb=no


a=rtpmap:101 telephone-event/8000


a=fmtp:101 0-15


a=ptime:20


a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt


a=sendrecv


SIP/2.0 100 Trying


Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport=2066;received=24.119.220.154


From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz


To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a


Call-ID: 52b0f3c5d925-dsalfsaxbvip


CSeq: 1 INVITE


Content-Length: 0




SIP/2.0 200 Ok


Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport=2066;received=24.119.220.154


From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz


To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a


Call-ID: 52b0f3c5d925-dsalfsaxbvip


CSeq: 1 INVITE


Contact: <sip:112@24.119.220.155:5061;transport=tls>


Supported: 100rel, replaces, norefersub


Allow-Events: refer


Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE


Accept: application/sdp


User-Agent: Vodia-PBX/5.1.3


Content-Type: application/sdp


Content-Length: 348




v=0


o=- 677813057 677813057 IN IP4 24.119.220.155


s=-


c=IN IP4 24.119.220.155


t=0 0


m=audio 19170 RTP/AVP 9 0 8 101


a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:KIovh3hlKqgGlVvULirklht4JS79+jXzdtOGGeMr


a=rtpmap:9 G722/8000


a=rtpmap:0 PCMU/8000


a=rtpmap:8 PCMA/8000


a=rtpmap:101 telephone-event/8000


a=fmtp:101 0-16


a=ptime:20


a=sendrecv


ACK sip:112@24.119.220.155:5061;transport=tls SIP/2.0


Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-xbj4q151mgkb;rport


From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz


To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a


Call-ID: 52b0f3c5d925-dsalfsaxbvip


CSeq: 1 ACK


Max-Forwards: 70


Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1


Proxy-Require: buttons-snom320


Content-Length: 0




BYE sip:112@24.119.220.155:5061;transport=tls SIP/2.0


Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-x0tq5z5vo8lp;rport


From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz


To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a


Call-ID: 52b0f3c5d925-dsalfsaxbvip


CSeq: 2 BYE


Max-Forwards: 70


Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1


User-Agent: snom320/8.7.3.25


RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0


RTP-TxStat: Total_Tx_Pkts=502,Tx_Pkts=502,Remote_Tx_Pkts=0


Proxy-Require: buttons-snom320


Content-Length: 0




SIP/2.0 200 Ok


Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-x0tq5z5vo8lp;rport=2066;received=24.119.220.154


From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz


To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a


Call-ID: 52b0f3c5d925-dsalfsaxbvip


CSeq: 2 BYE


Contact: <sip:112@24.119.220.155:5061;transport=tls>


User-Agent: Vodia-PBX/5.1.3


Content-Length: 0
Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

 Share

×
×
  • Create New...