Mailing List Archive

Early media with cisco 5400
Hello,



I have a cisco 5400 and I have a problem with early media.



When I receive a call from PSTN to voip and then this call is forwarded to a
PSTN destination via an other GW, this other GW send me back a 183 Session
in progress with SDP .



The problem is that the GW is not transferring the voice to the PSTN or as
I use a RTP proxy, maybe the cisco AS5400 don't send RTP traffic, so the RTP
proxy can't start the relay (he wait 1 RTP packet to detect the
publicIP/port)



When the user answer , the tha call is OK in both way.



any idea how to solve this ?





In my config I have set:





voice call send-alert

voice rtp send-recv

!



dial-peer voice 2 voip

destination-pattern 2T

progress_ind setup enable 3

progress_ind progress enable 8

translate-outgoing called 1

voice-class codec 7

session protocol sipv2

session target ipv4:203.40.5.100

dtmf-relay rtp-nte

no vad

!





gateway

!

sip-ua

nat symmetric check-media-src

retry invite 8

retry bye 2

retry cancel 2

timers expires 120000





link to 2 calls, first only 183 session in progress and secondl, an answered
calls:



http://pastebin.com/m26a0e423
Re: Early media with cisco 5400 [ In reply to ]
Hello all,



I do a test without RTP proxy to check if it’s the problem.



To do this test I use same cisco GW for the incoming and outgoing call, and
I have exactly the same problem.





Laurent



De : Laurent Schweizer [mailto:laurent.schweizer@peoplefone.com]
Envoyé : samedi 25 juillet 2009 18:01
À : cisco-voip@puck.nether.net
Objet : [cisco-voip] Early media with cisco 5400





Hello,



I have a cisco 5400 and I have a problem with early media.



When I receive a call from PSTN to voip and then this call is forwarded to a
PSTN destination via an other GW, this other GW send me back a 183 Session
in progress with SDP .



The problem is that the GW is not transferring the voice to the PSTN or as
I use a RTP proxy, maybe the cisco AS5400 don’t send RTP traffic, so the RTP
proxy can’t start the relay (he wait 1 RTP packet to detect the
publicIP/port)



When the user answer , the tha call is OK in both way.



any idea how to solve this ?





In my config I have set:





voice call send-alert

voice rtp send-recv

!



dial-peer voice 2 voip

destination-pattern 2T

progress_ind setup enable 3

progress_ind progress enable 8

translate-outgoing called 1

voice-class codec 7

session protocol sipv2

session target ipv4:203.40.5.100

dtmf-relay rtp-nte

no vad

!





gateway

!

sip-ua

nat symmetric check-media-src

retry invite 8

retry bye 2

retry cancel 2

timers expires 120000





link to 2 calls, first only 183 session in progress and secondl, an answered
calls:



http://pastebin.com/m26a0e423