[asterisk-users] Nat issue one way audio on IP dial

Jim Dickenson dickenson at cfmc.com
Wed Jul 28 11:36:51 CDT 2010 

Do you have your softphone setup to use a stun server so it can send it's public IP address in the SIP packets? I see in the SIP debug output a 192.168 address for the RTP packets to go to which of course will not work.
-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/



On Jul 28, 2010, at 9:23 AM, Nasir Javaid wrote:

> hi there,
> 
> i have posted earlier on the list but got no satisfying answer. the problem is not big.
> 
> I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone.
> 
> Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can hear the called user but the called user can not here the caller voice.
> 
> If the caller calls the other user by username instead of IP:Port , the voice is perfect both ways. 
> 
> what i have noticed is that IP:Port dial is missing a parameter "rinstance" in "Contact" , "To" headers for adf. what is "rinstance" for? Also something with "Contact" header seems fishy. or RTP issue.
>  
> that is 
> 
> Dial(SIP/adf,30,r) works fine with bothway audio, but
> 
> Dial(SIP/116.18.35.235:28614,30,r) has one way audio.
>             /                                 \
>             |                                  |            
>              this is IP:Port of of adf
> 
> please help as it's almost 2 weeks and i have found to suitable answer from any forum. I nead to know what can i do to modify Headers or settings in conf files to correct this problem.
> 
> Below is the conf of calling user
> 
> [pepsi]
> username=pepsi
> type=friend
> secret=123456
> qualify=yes
> nat=no
> insecure=port,invite
> incominglimit=1
> outgoinglimit=1
> host=dynamic
> dtmfmode=rfc2833
> context=out
> canreinvite=yes
> callerid="pepsi coke" <12345678901>
> accountcode=6:0:pepsi
> amaflags=default
> disallow=all
> allow=alaw
> allow=ulaw
> allow=g729
> allow=gsm
> 
> Below is the conf of called user
> 
> [adf]
> username=adf
> type=friend
> secret=123456
> qualify=yes
> nat=yes
> insecure=port,invite
> incominglimit=2
> outgoinglimit=2
> host=dynamic
> dtmfmode=rfc2833
> context=user
> canreinvite=yes
> callerid="adf xyz" <11223344556>
> accountcode=1:0:adf
> amaflags=default
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> allow=gsm
> 
> 
> 
> below is my sip debug after dialing
> 
> Audio is at 79.80.x.x port 16238
> Adding codec 0x8 (alaw) to SDP
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (NAT) to 116.18.35.235:28614:
> INVITE sip:adf at 116.18.35.235:28614 SIP/2.0
> Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport
> From: "pepsi coke" <sip:12345678901 at 79.80.x.x:5678>;tag=as42ec768c
> To: <sip:adf at 116.18.35.235:28614>
> Contact: <sip:12345678901 at 79.80.x.x:5678>
> Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 21 Jul 2010 15:10:22 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 285
> 
> v=0
> o=root 9626 9626 IN IP4 79.80.x.x
> s=session
> c=IN IP4 79.80.x.x
> t=0 0
> m=audio 16238 RTP/AVP 8 0 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> 
> ---
> [Jul 21 11:10:22] WARNING[23814]: chan_sip.c:2872 sip_call: Setting auto-congest time to 15000 ms.
>     -- Called adf at 116.18.35.235:28614
> <------------>
> ast-server*CLI> 
> <--- SIP read from 116.18.35.235:28614 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678
> Contact: <sip:adf at 116.18.35.235:28614>
> To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
> From: "pepsi coke"<sip:12345678901 at 79.80.x.x:5678>;tag=as42ec768c
> Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> CSeq: 102 INVITE
> User-Agent: X-Lite release 1104o stamp 56125
> Content-Length: 0
> 
> 
> <------------->
> --- (9 headers 0 lines) ---
>     -- SIP/116.18.35.235:28614-007f4660 is ringing
> ast-server*CLI> 
> <--- SIP read from 116.18.35.235:28614 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678
> Contact: <sip:adf at 116.18.35.235:28614>
> To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
> From: "pepsi coke"<sip:12345678901 at 79.80.x.x:5678>;tag=as42ec768c
> Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
> Content-Type: application/sdp
> User-Agent: X-Lite release 1104o stamp 56125
> Content-Length: 185
> 
> v=0
> o=- 6 2 IN IP4 192.168.0.12
> s=CounterPath X-Lite 3.0
> c=IN IP4 192.168.0.12
> t=0 0
> m=audio 55246 RTP/AVP 8 0 101
> a=fmtp:101 0-15
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
> 
> <------------->
> --- (11 headers 9 lines) ---
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP audio format 101
> Peer audio RTP is at port 192.168.0.12:55246
> Found description format telephone-event for ID 101
> Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
> Peer audio RTP is at port 192.168.0.12:55246
> list_route: hop: <sip:adf at 116.18.35.235:28614>
> [Jul 21 11:10:27] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing enforced for session 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> set_destination: Parsing <sip:adf at 116.18.35.235:28614> for address/port to send to
> set_destination: set destination to 116.18.35.235, port 28614
> Transmitting (NAT) to 116.18.35.235:28614:
> ACK sip:adf at 116.18.35.235:28614 SIP/2.0
> Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK07eb06b5;rport
> From: "pepsi coke" <sip:12345678901 at 79.80.x.x:5678>;tag=as42ec768c
> To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
> Contact: <sip:12345678901 at 79.80.x.x:5678>
> Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
> 
> 
> ---
>     -- Call on SIP/116.18.35.235:28614-007f4660 left from hold
>     -- SIP/116.18.35.235:28614-007f4660 answered SIP/pepsi-9fdfb9a0
> ast-server*CLI> 
> <--- SIP read from 116.18.35.235:28614 --->
> 
> 
> 
> <------------->
> --- (0 headers 1 lines) ---
> ast-server*CLI> 
> <--- SIP read from 116.18.35.235:28614 --->
> 
> 
> 
> <------------->
> --- (0 headers 1 lines) ---
> Scheduling destruction of SIP dialog '0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x' in 32000 ms (Method: INVITE)
> [Jul 21 11:11:03] DEBUG[23814]: chan_sip.c:5695 reqprep: Strict routing enforced for session 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> set_destination: Parsing <sip:adf at 116.18.35.235:28614> for address/port to send to
> set_destination: set destination to 116.18.35.235, port 28614
> Reliably Transmitting (NAT) to 116.18.35.235:28614:
> BYE sip:adf at 116.18.35.235:28614 SIP/2.0
> Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK36df65b5;rport
> From: "pepsi coke" <sip:12345678901 at 79.80.x.x:5678>;tag=as42ec768c
> To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c
> Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
> 
> 
> 
> 
> Nasir Javaid
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100728/2e4d9106/attachment.htm

More information about the asterisk-users mailing list