[Asterisk-Users] RTP Timestamp errors

[Erik]

Hi list,

I know * generates it's outgoing RTP stream based on the incomming RTP stream, i'm having some audio problems after i recieve an rtp reinvite from my
carrier.

Situation:

Phone -- Asterisk A --- Asterisk B --- Carrier --- PSTN

Asterisk A: reinvite = no
Asterisk B: reinvite = no

If i dial out on phone via asterisk A, Asterisk B relay's the INVITE to the carrier, after the PSTN answers the call the carrier sends a reinvite to
Asterisk B to change the ip to one of the the Media Gateways of the carrier, the media gateway however sends RTP with a completely different timestamp
to Asterisk B, so Asterisk B copies that timstamp and Asterisk A gets an audio hickup.

IE

asterisk B recieves:                                           asterisk B sends to A
sequence 1 timestamp  0 SSRC 1234 from ip 1.2.3.4              sequence 1 timestamp 0 SSRC 4321 from ip Asterisk B
sequence 2 timestamp 30 SSRC 1234 from ip 1.2.3.4              sequence 2 timestamp 30 SSRC 4321 from ip Asterisk B
sequence 3 timestamp 60 SSRC 1234 from ip 1.2.3.4              sequence 3 timestamp 60 SSRC 4321 from ip Asterisk B

so far, so good, but then Asterisk B recieves a reinvite from the carrier and start to send this rtp to Asterisk A

sequence 500 timestamp 500000 SSRC 5678 from ip 1.2.3.4              sequence 4 timestamp 500000 SSRC 1234 from ip Asterisk B
sequence 501 timestamp 500030 SSRC 5678 from ip 1.2.3.4              sequence 5 timestamp 500030 SSRC 1234 from ip Asterisk B
sequence 502 timestamp 500060 SSRC 5678 from ip 1.2.3.4              sequence 6 timestamp 500060 SSRC 1234 from ip Asterisk B

IMHO Asterisk B should change it SSRC to tell Asterisk A the RTP source has changed (and fix this timestamp gap)?


Erik