SIP TO SIP on LAN no voice

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gielchina
Posts: 7
Joined: Tue Aug 28, 2012 5:12 am

SIP TO SIP on LAN no voice

Postby gielchina » Tue Mar 18, 2014 7:26 am

Dear friends,

I am using two sip phone on same LAM network and trying to call between. There is no voice, not even one way. I am giving output of console, can anyone check where is the problem.

Monitor 06:29:58:583: basetype=console, ipaddress=*, user=gielchina, event=*, request=*, serveripaddress=*, server=*, regex=.*.
DialPlan 06:30:01:412 sip1(9588): New call from udp:14.147.55.3:11454 successfully authenticated by digest.
DialPlan 06:30:01:443 sip1(9588): Using dialplan Outging Basic for Out call to sip:101@sipsorcery.com.
NewCall 06:30:01:474 sip1(9588): Executing script dial plan for call to 101.
DialPlan 06:30:01:490 sip1(9588): New outgoing call received to 101.
DialPlan 06:30:01:490 sip1(9588): Commencing Dial with: choffice@sipsorcery.com[swln=China Office].
DialPlan 06:30:01:505 sip1(9588): Call leg is for local domain looking up bindings for choffice@sipsorcery.com for call leg choffice@sipsorcery.com.
DialPlan 06:30:01:505 sip1(9588): 1 found for choffice@sipsorcery.com.
DialPlan 06:30:01:505 sip1(9588): ForkCall commencing call leg to sip:choffice@14.147.55.3:9689.
DialPlan 06:30:01:505 sip1(9588): SIPClientUserAgent Call using alternate outbound proxy of udp:67.222.131.147:5060.
DialPlan 06:30:01:505 sip1(9588): Switching to sip:choffice@14.147.55.3:9689 via udp:67.222.131.147:5060.
DialPlan 06:30:01:505 sip1(9588): SDP on UAC call was set to NOT mangle, RTP socket 172.16.1.101:47190.
DialPlan 06:30:02:318 sip1(9588): Information response 100 Trying for sip:choffice@14.147.55.3:9689.
DialPlan 06:30:02:334 sip1(9588): Information response 180 Ringing for sip:choffice@14.147.55.3:9689.
DialPlan 06:30:02:334 sip1(9588): UAS call progressing with Ringing.
DialPlan 06:30:02:521 sip1(9588): Information response 180 Ringing for sip:choffice@14.147.55.3:9689.
DialPlan 06:30:02:521 sip1(9588): UAS call ignoring progress response with status of 180 as already in Proceeding.
NATKeepAlive 06:30:03:599 sip1(19928): Requesting NAT keep-alive from proxy socket udp:67.222.131.147:5060 to udp:14.147.55.3:11454.
DialPlan 06:30:04:459 sip1(9588): Response 200 OK for sip:choffice@14.147.55.3:9689.
DialPlan 06:30:04:459 sip1(9588): SDP on UAC response was set to NOT mangle, RTP socket 14.147.55.3:14330.
DialPlan 06:30:04:459 sip1(9588): Cancelling all call legs for ForkCall app.
DialPlan 06:30:04:459 sip1(9588): Answering client call with a response status of 200.
DialPlan 06:30:04:505 sip1(9588): Dial command was successfully answered in 3.00s.
DialPlan 06:30:04:505 sip1(9588): Dialplan cleanup for gielchina.
DialPlan 06:30:04:584 sip1(9588): Dial plan execution completed with normal clearing.
NATKeepAlive 06:30:08:693 sip1(19928): Requesting NAT keep-alive from proxy socket udp:67.222.131.147:5060 to udp:14.147.55.3:9689.

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Aaron
Site Admin
Posts: 4550
Joined: Thu Jul 12, 2007 12:13 am

Re: SIP TO SIP on LAN no voice

Postby Aaron » Tue Mar 18, 2014 8:38 am

You need to find a way to get your SIP devices to use their private IP addresses rather than them substituting their public one.

Code: Select all

DialPlan 06:30:01:505 sip1(9588): SDP on UAC call was set to NOT mangle, RTP socket 172.16.1.101:47190.
...
DialPlan 06:30:04:459 sip1(9588): SDP on UAC response was set to NOT mangle, RTP socket 14.147.55.3:14330.


What those two lines tell you is that the the attempt will be made to set up the audio stream between sockets 172.16.1.101:47190 and 14.147.55.3:14330. The first IP address is from your private LAN and is fine for this case however the second IP address is the public IP address of your network and normally won't work properly for devices on your private LAN.

One possible way to get the second device to use its private IP address is to turn off any STUN setting or blank out the STUN server if it's set.


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