Re: [OpenSIPS-Users] One Way Audio Using X-Lite, OpenSIPS & Asterisk (Ross Beer)

2009-10-23 Thread Ross Beer 


Hi,
 
Here is the sip trace for the calls, it strange as all other phone work except 
X-Lite, it appears that sound is not getting from the softphone to the server. 
Though if the softphone talks directly to asterisk it works ok. This is the 
case if MediaProxy is used or not.
 
There is a lack of codecs in the invite which is strange as I have 4 enabled on 
the server and the softphone.
 
Thank you for your help!
 
Ross
 
--
 
INVITE sip:160@ SIP/2.0
Via: SIP/2.0/UDP :9302;branch=z9hG4bK-d8754z-4e352701ef65e42a-1---d8754z-;rport
Max-Forwards: 69
Contact: :9302>
To: "160">
From: "Ross">;tag=ad038800
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 236
v=0
o=- 0 2 IN IP4 192.168.1.222
s=CounterPath X-Lite 3.0
c=IN IP4 
t=0 0
m=audio 10006 RTP/AVP 0 101
a=alt:1 1 : 9ImKPFaa h9RvD+sl 192.168.1.222 10006
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP ;branch=z9hG4bKfe32.d0a3adc2.0;received=
Via: SIP/2.0/UDP :9302;received=;branch=z9hG4bK-d8754z-4e352701ef65e42a-1---d8754z-;rport=9302
From: "Ross">;tag=ad038800
To: "160">;tag=as57033d07
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 1 INVITE
User-Agent: Asterisk 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="", nonce="6d81b1ec"
Content-Length: 0
---
ACK sip:160@ SIP/2.0
Via: SIP/2.0/UDP :9302;branch=z9hG4bK-d8754z-4e352701ef65e42a-1---d8754z-;rport
To: "160">;tag=as57033d07
From: "Ross">;tag=ad038800
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 1 ACK
Content-Length: 0
---
INVITE sip:160@ SIP/2.0
Via: SIP/2.0/UDP :9302;branch=z9hG4bK-d8754z-063f0a0c111bba4a-1---d8754z-;rport
Max-Forwards: 69
Contact: :9302>
To: "160">
From: "Ross">;tag=ad038800
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest 
username="10002*200",realm="",nonce="6d81b1ec",uri="sip:160@",response="cbb367ca716a5e7c62ff962824996533",algorithm=MD5
Content-Length: 236
v=0
o=- 0 2 IN IP4 192.168.1.222
s=CounterPath X-Lite 3.0
c=IN IP4 
t=0 0
m=audio 10006 RTP/AVP 0 101
a=alt:1 1 : 9ImKPFaa h9RvD+sl 192.168.1.222 10006
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
-
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ;branch=z9hG4bKce32.ba4f096.0;received=
Via: SIP/2.0/UDP :9302;received=;branch=z9hG4bK-d8754z-063f0a0c111bba4a-1---d8754z-;rport=9302
Record-Route: ;lr=on;ftag=ad038800>
From: "Ross">;tag=ad038800
To: "160">
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 2 INVITE
User-Agent: Asterisk 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: :5061>
Content-Length: 0
--
SIP/2.0 200 OK
Via: SIP/2.0/UDP ;branch=z9hG4bKce32.ba4f096.0;received=
Via: SIP/2.0/UDP :9302;received=;branch=z9hG4bK-d8754z-063f0a0c111bba4a-1---d8754z-;rport=9302
Record-Route: ;lr=on;ftag=ad038800>
From: "Ross">;tag=ad038800
To: "160">;tag=as0f58ca6e
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 2 INVITE
User-Agent: Asterisk 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: :5061>
Content-Type: application/sdp
Content-Length: 272
v=0
o=root 1609040672 1609040672 IN IP4 
s=Asterisk PBX 1.6.0.16-rc2
c=IN IP4 
t=0 0
m=audio 49356 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

ACK sip:160@:5061 SIP/2.0
Via: SIP/2.0/UDP :9302;branch=z9hG4bK-d8754z-4b605e0b6855e57e-1---d8754z-;rport
Max-Forwards: 69
Route: ;lr;ftag=ad038800>
Contact: :9302>
To: "160">;tag=as0f58ca6e
From: "Ross">;tag=ad038800
Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
CSeq: 2 ACK
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest 
username="10002*200",realm="",nonce="6d81b1ec",uri="sip:160@",response="cbb367ca716a5e7c62ff962824996533",algorithm=MD5
Con

Re: [OpenSIPS-Users] One Way Audio Using X-Lite, OpenSIPS & Asterisk (Ross Beer)

2009-10-23 Thread Raúl Alexis Betancor Santana 
On Friday 23 October 2009 14:19:45 Ross Beer wrote:
> Hi,
>
> Here is the sip trace for the calls, it strange as all other phone work
> except X-Lite, it appears that sound is not getting from the softphone to
> the server. Though if the softphone talks directly to asterisk it works ok.
> This is the case if MediaProxy is used or not.
>
> There is a lack of codecs in the invite which is strange as I have 4
> enabled on the server and the softphone.

Seems you don't look on the wright place .. because your X-Lite is offering 
PCMU ... just look at the m= line on the first invite.


> INVITE sip:160@ SIP/2.0
> Via: SIP/2.0/UDP  ADDRESS>:9302;branch=z9hG4bK-d8754z-4e352701ef65e42a-1---d8754z-;rport
> Max-Forwards: 69
> Contact: :9302>
> To: "160">
> From: "Ross">;tag=ad038800
> Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM.
> CSeq: 1 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO Content-Type: application/sdp
> User-Agent: X-Lite release 1103k stamp 53621
> Content-Length: 236
> v=0
> o=- 0 2 IN IP4 192.168.1.222
> s=CounterPath X-Lite 3.0
> c=IN IP4 
> t=0 0
> m=audio 10006 RTP/AVP 0 101
> a=alt:1 1 : 9ImKPFaa h9RvD+sl 192.168.1.222 10006
> a=fmtp:101 0-15
> a=rtpmap:101 telephone-event/8000
> a=sendrecv

Umm ... umm ... by ROUTER IP ADDRESS do you mean your public IP on your NAT 
router ? ...
does your router have SIP-ALG activated ?, that could explain the problem.
This INVITE is the one that just arrive at your proxy? ... if so .. your NAT 
router is doing sip-alg .. so mangling all the SIP dialog, and they usually 
does a very bad job with that.


-- 
Raúl Alexis Betancor Santana
Dimensión Virtual

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Re: [OpenSIPS-Users] One Way Audio Using X-Lite, OpenSIPS & Asterisk (Ross Beer)

2009-10-23 Thread Saúl Ibarra 
Do you have 'discover global IP' and 'use ICE' setting enabled in
X-lite? I remember I had some issues in the past with that... let your
proxy handle everything for you.


-- 
/Saúl
http://www.saghul.net | http://www.sipdoc.net

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