[cisco-voip] RTP only one way?

[Gerd Feiner]

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
-------------- next part --------------
A non-text attachment was scrubbed...
Name: PGP.sig
Type: application/pgp-signature
Size: 186 bytes
Desc: Signierter Teil der Nachricht
Url : https://puck.nether.net/pipermail/cisco-voip/attachments/20050429/34debd37/PGP.bin