[cisco-voip] RTP only one way?

Gerd Feiner g.feiner at cablesurf.de
Fri Apr 29 06:49:51 EDT 2005

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
   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
  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
  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.

Gerd Feiner
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