Dynamic codecs are indicated by the a=rtpmap lines. Those lines may the number used on m=audio to the description of the codec. All static payload types *may* be overriden to another codec using a=rtpmap lines, but usually aren't. If they're not overridden, they use the values on the page you already found. Note a=rtpmap lines are optional for static payload types. That does cause problems on SIP clients that failed to read that part of the standard, so for those set verbose_sdp=true on the channel. That then means your SDP is larger and may give problems where packets are too large and get fragmented, which SIP/UDP doesn't handle very well. -Steve On 23 June 2013 04:23, Nathan Neulinger <nneul@mst.edu> wrote:
I think this may be what you're looking for...
http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml
96+ are dynamic apparently
0, 8 pcmu and pcma 3 is gsm
At least according to that list... I don't know enough about the protocol to know how the dynamic ones are assigned.
This that's odd though is that the list on that page is in decimal, with comfort noise at 13. That would make sense if it were in hex (19), so maybe I'm misinterpreting.
-- Nathan
On 06/22/2013 10:02 PM, Sayyed Mohammad Emami Razavi wrote:
m=audio 5016 RTP/AVP 96 3 107 110 0 8 101. Is there any digits represents sip codecs in subject?!
Can any body describe this statement?!
For example in response to one of my questions, Steven told me: "Your phone is not offering any codecs in the INVITE: m=audio 15696 RTP/AVP 19 The only payload type listed is 19 (comfort noise) - there are no codecs listed." base on my logs: http://pastebin.freeswitch.org/21084
-- ------------------------------------------------------------ Nathan Neulinger nneul@mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect
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