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

[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@<SERVER IP ADDRESS> SIP/2.0
> Via: SIP/2.0/UDP <ROUTER IP
> ADDRESS>:9302;branch=z9hG4bK-d8754z-4e352701ef65e42a-1---d8754z-;rport
> Max-Forwards: 69
> Contact: <sip:10002*200@<ROUTER IP ADDRESS>:9302>
> To: "160"<sip:160@<SERVER IP ADDRESS>>
> From: "Ross"<sip:10002*200@<SERVER IP ADDRESS>>;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 <ROUTER IP ADDRESS>
> 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