You can use Wireshark to capture all IP packets, and then decode & play received and sent RTP (if its in one of the supported formats only G711 as far as I know).
You can access the RTP replay feature from the wireshark statistics / VOIP calls menu. stipus De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Paulo Vicentini Envoyé : lundi 27 octobre 2008 20:53 À : [email protected] Objet : [sipxtapi-dev] stop to playback decoded RTP I am facing a problem in a specific scenario: sipxtapiEP----------------media server plays message-------------------EP (several kinds) (both EPs are able to hear the message) sipxtapiEP(can't hear other EP ) ------------(message is over)------------------EP (is able to hear sipxtapiEP) Whenever an ingoing / outgoing call is intercepted by a media server that plays a message (e.g announcement / collect call message) before bridging between the finals endpoints I have the following problem: While playing the message both endpoints are able to hear the message but after all the message is played, sipxtapi EP is mute (can't hear other end EP), although it is able to send / receive RTP. The other end is able to hear (audio voice) sipxtapi EP after message is played. SDP session description remains the same by the end of the announcements (no re-invite). Without a media server (announcement) in mid of the call all is fine (receive / make). I do check NetInTask / MprFromNet and I saw that RTP packets are still been pushed after the end of the message. I will check Spk Task and others to see what's going on but any help to figure out my issue is welcome. Thank you
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