[cisco-voip] RTP only one way?

Kevin Thorngren kevint at cisco.com
Fri Apr 29 09:47:15 EDT 2005


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
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