[cisco-voip] RTP only one way?

[Gerd Feiner]

Hi Ryan,

the AS has only one IP-address - and RTP from _ATA_ to AS is working  
fine, but I can't hear the other party on the SIP-phone.

Brgds,
Gerd Feiner

Am 29.04.2005 um 18:48 schrieb Ryan Ratliff:

> I don't know how this applies to SIP but if your AS has more than one  
> IP address you should configure 'h323 bind voip srcaddr' under the  
> appropriate interface.  Could very well be that the AS is telling the  
> SIP device to send RTP to an IP address it can't route to.
>
> HTH
>
> -Ryan
> On Apr 29, 2005, at 11:59 AM, Gerd Feiner wrote:
>
> Hi Kevin,
>
> no there aren't any routing issues.  The AS can ping the sipura and  
> there are indeed RTP-packets from the AS to the sipura - about 1 every  
> two seconds, while there are many many packtes from the sipura to the  
> AS5350 ... when debugging SIP the AS5350 also tells about opening a  
> recv-only audio-stream:
>
> Apr 29 12:41:56 x.x.x.x 42940: Apr 29 10:48:30.444: sipSPIAddStream:  
> AddStream in idle state to open a 'recvonly' media session
>
> this is why I digged into and found that rtp send-receive command and  
> its exactly what happens:  voip can speak and is heard on pstn, but  
> not vice versa.
>
> any ideas?
>
> Brgds
> Gerd
>
> Am 29.04.2005 um 15:47 schrieb Kevin Thorngren:
>
>> Hi Gerd,
>>
>> Typically one way voice issues are due to IP Routing issues.  Can the  
>> 5300 ping the SIP Phone?
>>
>> This URL should help you diagnose the problem.
>> http://www.cisco.com/en/US/tech/tk652/tk698/ 
>> technologies_tech_note09186a008009484b.shtml
>>
>> Kevin
>> On Apr 29, 2005, at 6:49 AM, Gerd Feiner wrote:
>>
>>> Hi there,
>>>
>>> we have an AS5350 and using as a SIP-Gateway to the PSTN.  Now,  
>>> there is an intriguing issue:  SIP -> PSTN voice is audible, and  
>>> there is an RTP stream from the SIP-device - SIPURA - to the  
>>> mediagateway.  But there is no stream in the SIP direction coming  
>>> from the AS5350.  I already found the
>>>
>>> voice rtp send-receive
>>>
>>> command - but it didn't do the trick. As of now, I wasn't able to  
>>> ascertain the source of the problem.  It doesn't matter who is  
>>> initiating the call, it's always the same effect.
>>>
>>> Don't know which part of our config you need, but here are a few:
>>>
>>> voice rtp send-recv
>>> !
>>> voice service pots
>>>  fax protocol pass-through g711alaw
>>> !
>>> voice service voip
>>>  signaling forward rawmsg
>>>  fax protocol pass-through g711alaw
>>>  sip
>>>   rel1xx disable
>>>   no call service stop
>>> !
>>> ...
>>> !
>>> interface Serial3/0:15
>>>  no ip address
>>>  isdn switch-type primary-net5
>>>  isdn incoming-voice modem
>>>  isdn sending-complete
>>>  no cdp enable
>>> !
>>> voice-port 3/0:D
>>>  bearer-cap Speech
>>> !
>>> dial-peer voice 1 pots
>>>  tone ringback alert-no-PI
>>>  application session
>>>  incoming called-number 143677..
>>>  destination-pattern .
>>>  translate-outgoing calling 20
>>>  translate-outgoing called 20
>>>  supplementary-service pass-through
>>>  no digit-strip
>>>  direct-inward-dial
>>>  port 3/0:D
>>> !
>>> dial-peer voice 2 voip
>>>  tone ringback alert-no-PI
>>>  application session
>>>  incoming called-number .
>>>  destination-pattern 143677..
>>>  voice-class codec 10
>>>  session protocol sipv2
>>>  session target ipv4:x.x.x.x
>>>  supplementary-service pass-through
>>> !
>>> !
>>> dial-peer search type voice data
>>> sip-ua
>>>  nat symmetric check-media-src
>>>  sip-server ipv4:x.x.x.x
>>>
>>> this isn't by far complete, but it seems to be the important part as  
>>> I figured.  In addition, I don't really understand all of the  
>>> commands set, most of it was from an example, part is from the  
>>> ?-help system and another part is from cisco's voice config guide  
>>> ...
>>>
>>> Glad if someone could help.
>>>
>>> Brgds,
>>> Gerd Feiner
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
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