[asterisk-users] "remote" SIP, no audio, or one way audio.
dave cantera
david.cantera at iacnet.net
Sat Apr 7 12:53:02 MST 2007
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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