[asterisk-users] Asterisk chan_sip removing ptime from DSP?

[Benoît Panizzon]

Hi List

Just ran into another weird issue...

In Swiss Telephone Interconnection, ptime=20 is a requirement.

So on our SBC we enforce the presence of ptime=20 to avoid issues.

I have an asterisk with chan_sip in the LAB which behaves weirdly...

Inbound SDP audio part:

m=audio 15542 RTP/AVP 9 8 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:4hJMPvZmRA03KxLg8Hp+aHqeIhnmYBSQtwlT+Vkr a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Outbound SDP audio part

m=audio 18802 RTP/AVP 8 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

Why is ptime missing on the outbound leg?

Our SBC answers 415

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