Asterisk chan_sip removing ptime from DSP?
Benoît Panizzon <[email protected]>
| Newsgroups | gmane.comp.telephony.pbx.asterisk.user |
|---|---|
| Organization | ImproWare AG |
| Message-ID | <[email protected]> |
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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Mit freundlichen Grüssen
-Benoît Panizzon- @ HomeOffice und normal erreichbar
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I m p r o W a r e A G - Leiter Commerce Kunden
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Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz Web http://www.imp.ch
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