I have tried setting up a basic VoIP configuration between 2 branch sites using 2 service provider networks using Cisco 1750 routers.
One setup was over a frame relay network and the other over a PPTP tunnel link that goes through several intermediate routers.
The setup worked fine on the frame relay n/w. Over the PPTP link however,the calling party doesn't get the ringback although the phone at the other end rings. When the called party answers, he can hear the calling party but the latter doesn't receive the voice stream. This was always the case.
With some debugging I found that the called party sends the RTP voice stream to the wrong RTP port at the calling party. The port nos were always 2 greater than the correct one.
Eg:- called party sends the voice stream to 18002 when the calling party's end shows that its RTP port should actually be 18000. There was no port mismatch the other way.
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