[asterisk-users] "remote" SIP, no audio, or one way audio.

[Joe Acquisto]

I never get this far, apparently.   While the connection seems to be made, and calls can be "completed" (rings, answers) there is no audio.   On CLI, I can see what appears to be call being made and connected.  These are x-lite phones (for testing, one hopes) there appears to be no codec selection available.

I see no CODEC dialog.  What I see is six iterations of the below:

. . . .
---

Retransmitting #6 (NAT) to xx.xx.xx.xx:64909:
NOTIFY sip:3306 at xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport
From: "nnnnn"<sip:3306 at xx.xx.xx.xx;tag=as67e5c857
To: "nnnnn"<sip:3306 at xx.xx.xx.xx>;tag=9c58a77e
Contact: <sip:3306 at 192.168.0.xxx>
Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE.
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: terminated;reason=timeout
Content-Length: 0
-----

Does this imply anyting to anyone?

Call can be made, after this.

joe a.

******
dave cantera <david.cantera at iacnet.net> Wrote: 4/7/2007 3:53 PM:
> joe,
> when I have problems with audio and other connections seem to work, I 
> always look for a codec incompatibility...  use  'sip set debug peer 
> <extension>'  and look for the codec handshaking... make sure both 
> extensions have a compatible codec choice...
> daveC
> 
> Using INVITE request as basis request - 58867de69e90aa51 at 192.168.15.100 
> Found user '401'
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 3
> Found RTP video format 99
> Peer audio RTP is at port 192.168.15.100:5004
> 
> *Found description format PCMU for ID 0
> Found description format PCMA for ID 8
> Found description format GSM for ID 3
> Found description format H264 for ID 99
> 
> *Capabilities: us - 0x20000e (gsm|ulaw|alaw|h264), peer - 
> audio=0x20000e 
> (gsm|ulaw|alaw|h264)/video=0x200000 (h264), combined - 0x20000e 
> (gsm|ulaw|alaw|h264)
> 
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 
> (nothing), combined - 0x0 (nothing)
> Peer audio RTP is at port 192.168.15.100:5004
> Peer video RTP is at port 192.168.15.100:5006
> Looking for 404 in inbound-video (domain sip3701.ibsonecall.com)
> list_route: hop: <sip:401 at 192.168.15.100:5060;user=phone>
> 
> 
> 
> Joe Acquisto wrote:
>> Steve Totaro <stevetotaro at hotmail.com> Wrote: 4/4/2007 8:44 PM:
>>   
>>> Joe Acquisto wrote:
>>>     
>>>> Attempts to do SIP thru firewall (IPCop) are unsuccessful.  using x-lite 
>>>> softphones, for eval/testing.  They do get registered, and can call each 
>>>> other, but mostly get no audio, sometimes one way audio.
>>>>
>>>> Suggestions/fixes?
>>>>
>>>> joe a.
>>>>   
>>>>       
>>> Is there NAT on both sides?  Are you using qualify?  Paint a clearer 
>>> picture.
>>>
>>>     
>>
>>
>> Sorry, I missed your reply, till now.
>>
>> ------------------switch
>>      |      |     |----phones
>>      |      |---------asterisk box
>>      
>> |---------------IPcop------------|---internet-----|-----home/remote-office--
>> --|----sip phone
>>                                                                             
>> |-----ditto
>>
>> Hope that is intelligible.
>>
>> joe a
>>
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>>   
> 
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