[Asterisk-Dev] mulitpart mixed application sdp

Jerry Geis geisj at pagestation.com
Thu Jun 2 14:51:49 MST 2005

I have modified the source to (chan_sip.c) to not return on error of the multipart sip.
I have inserted manually the values for "m = " and "c = " that I got from the debug output.
I am very close to getting something to work.
The problem is I dont ALWAYS get audio.

AS you can see below audio formats are found.

I posted to this list as I thought users would not be technical enough for finding a solution
to connecting to the nortel CS 1000.

ANy ideas on why I dont always get audio???

THanks for your thoughts.


Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
    -- Executing NoOp("SIP/SXNTM1SS1-5ea0", "2828") in new stack
    -- Executing Goto("SIP/SXNTM1SS1-5ea0", "default|s|1") in new stack
    -- Goto (default,s,1)
    -- Executing Wait("SIP/SXNTM1SS1-5ea0", "1") in new stack
    -- Executing Answer("SIP/SXNTM1SS1-5ea0", "") in new stack
    -- Executing Playback("SIP/SXNTM1SS1-5ea0", "demo-congrats") in new stack
    -- Playing 'demo-congrats' (language 'en')

Jerry Geis wrote:
...interesting SIP message by the way... If you could find out a bit
more about
Nortel's SIP implementation I would like some info...

>/ Sip read: INVITE
/>/ sip:2828;phone-context=cdp.udp at qg.com <http://lists.digium.com/mailman/listinfo/asterisk-dev>:5060;maddr=;transport=udp;user=phone;x-nt-redirect=redirect-server
/>/ SIP/2.0
/What is "x-nt-redirect"?

>/ From:
/>/ <sip:3173241052;phone-context=+1 at qg.com <http://lists.digium.com/mailman/listinfo/asterisk-dev>;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed
/>/ To: <sip:2828;phone-context=cdp.udp at qg.com <http://lists.digium.com/mailman/listinfo/asterisk-dev>;user=phone>
/>/ Call-ID: 103548e8-c22da8c0-13c4-429efa71-657b8da-517 at qg.com <http://lists.digium.com/mailman/listinfo/asterisk-dev>
/>/ CSeq: 1 INVITE
/>/ Via: SIP/2.0/UDP;branch=z9hG4bK-429efa71-657b8da-2522
/>/ Max-Forwards: 70
/>/ Supported: 100rel,sipvc,replaces
/Hmm - what is "sipvc" ?


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