Hi,
I have a generic Sip and RTP query.
I want to connect party A, B and C in such a way that C is sending RTP voice stream to A while A is sending RTP voice stream to B (but not C). In simple terms, a party is sending media stream to one IP:port but receiving it from some other IP:port combination as shown below:
|-------->--------- B [10.10.10.101:4010]
|
A [10.10.10.100:4000] --------------|
|
|--------<--------- C [10.10.10.102:4020]
I have setup the calls with required SDP modification to achieve this flow. Call Leg A-B is connected first and SDP received from A is sent to C to connect second Leg of A-C. The voice from A->B is flowing fine; I can listen at B end. However, voice stream sent by C is not getting received in A's machine. Wireshark on A's machine is not showing any RTP packets while Wireshark on C's machine is showing RTP sent out to correct IP and port of A's machine.
Can anyone suggest what could be the issue? Is it fine as per RTP spec to send to one IP:UDP port and receive from other?
I have verified that Voice Codec on both streams is fine i.e. PCM alaw. Also, RTP flows correctly when connecting a normal call between A and C.
Regards
Girish
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