[Asterisk-video] Lifesize HD X10 endpoints & Video Resolution and framerate configuration question

Sat Apr 16 23:46:13 CDT 2011

Compiled cleanly but no joy so far, possibly config my end, I couldn't
just add the chan_sip lib to the running server (asterisk now redhat,
complained about compile time options being different ;-/ )so I have a
very basic test server running at the mo.

I can't get either endpoint to connect:

[Apr 17 16:30:50] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio:
Got Siren14 offer at 32000 bps, but only 48000 bps supported;
ignoring.
[Apr 17 16:30:50] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio:
Got Siren14 offer at 24000 bps, but only 48000 bps supported;
ignoring.
[Apr 17 16:30:50] WARNING[5572]: chan_sip.c:8314 process_sdp:
Unsupported SDP media type in offer: application 60040 RTP/AVP 100
    -- Executing [2000 at sip:1] Dial("SIP/2001-00000000",
"SIP/2000,20,tr") in new stack
  == Using SIP RTP CoS mark 5
[Apr 17 16:30:50] WARNING[31184]: channel.c:1045 ast_best_codec: Don't
know any of 0x0 (nothing) formats
[Apr 17 16:30:50] WARNING[31184]: channel.c:1045 ast_best_codec: Don't
know any of 0x0 (nothing) formats
    -- Called 2000
    -- Nobody picked up in 20000 ms
    -- Auto fallthrough, channel 'SIP/2001-00000000' status is 'NOANSWER'
[Apr 17 16:31:22] WARNING[5572]: chan_sip.c:3386 retrans_pkt:
Retransmission timeout reached on transmission
5f76f24b5c4c98f5239e9f934f49ebd7 at 192.168.254.56:5060 for seqno 102
(Critical Request) -- See doc/sip-retransmit.txt.
Packet timed out after 32000ms with no response
  == Using SIP RTP CoS mark 5
[Apr 17 16:32:10] WARNING[32084]: chan_sip.c:8857 process_sdp_a_audio:
Got Siren14 offer at 32000 bps, but only 48000 bps supported;
ignoring.
[Apr 17 16:32:10] WARNING[32084]: chan_sip.c:8857 process_sdp_a_audio:
Got Siren14 offer at 24000 bps, but only 48000 bps supported;
ignoring.
[Apr 17 16:32:10] WARNING[32084]: chan_sip.c:8314 process_sdp:
Unsupported SDP media type in offer: control 60070 RTP/AVP 96
    -- Executing [2001 at sip:1] Dial("SIP/2000-00000002",
"SIP/2000,20,tr") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 2000
    -- Nobody picked up in 20000 ms
    -- Auto fallthrough, channel 'SIP/2000-00000002' status is 'NOANSWER'
[Apr 17 16:32:42] WARNING[5572]: chan_sip.c:3386 retrans_pkt:
Retransmission timeout reached on transmission
031126ca272c120d0fcee1002f5a1bc4 at 192.168.254.56:5060 for seqno 102
(Critical Request) -- See doc/sip-retransmit.txt.
Packet timed out after 32000ms with no response
  == Using SIP RTP CoS mark 5
[Apr 17 16:41:02] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio:
Got Siren14 offer at 32000 bps, but only 48000 bps supported;
ignoring.
[Apr 17 16:41:02] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio:
Got Siren14 offer at 24000 bps, but only 48000 bps supported;
ignoring.
[Apr 17 16:41:02] WARNING[5572]: chan_sip.c:8314 process_sdp:
Unsupported SDP media type in offer: application 60046 RTP/AVP 100
[Apr 17 16:41:02] NOTICE[5572]: chan_sip.c:21361
handle_request_invite: Call from '2001' to extension '*43' rejected
because extension not found in context 'sip'.
  == Using SIP RTP CoS mark 5
[Apr 17 16:41:04] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio:
Got Siren14 offer at 32000 bps, but only 48000 bps supported;
ignoring.
[Apr 17 16:41:04] WARNING[5572]: chan_sip.c:8857 process_sdp_a_audio:
Got Siren14 offer at 24000 bps, but only 48000 bps supported;
ignoring.
[Apr 17 16:41:04] WARNING[5572]: chan_sip.c:8314 process_sdp:
Unsupported SDP media type in offer: application 60052 RTP/AVP 100
[Apr 17 16:41:04] NOTICE[5572]: chan_sip.c:21361
handle_request_invite: Call from '2001' to extension '*43' rejected
because extension not found in context 'sip'.

On 16 April 2011 04:52, CM Rahman <cmrahman at yahoo.com> wrote:
> Gunnar,
>
> It compiled fine under both 1.6 and 1.8.  Now is the time to test it.
>
> Thanks
> CM
>
> ________________________________
> From: Gunnar Schaller <linux at nowin.de>
> To: Development discussion of video media support in Asterisk
> <asterisk-video at lists.digium.com>
> Sent: Fri, April 15, 2011 2:05:08 AM
> Subject: Re: [Asterisk-video] Lifesize HD X10 endpoints & Video Resolution
> and framerate configuration question
>
> Hello,
>
> Now I tried to compile it. Works! Attached my patch for 1.8 branch.
> For 1.6 you need to add the lines manually around line 10270 in
> chan_sip.c.
> What about JoelW? Did you try that patch?
>
> Regards,
> Gunnar
>
>
>
>> Gunnar,
>
>> I have tried to compile with the code you provided in 1.6.x
>
>> Here is what I get..
>
>  [CC] chan_sip.c ->> chan_sip.o
>> chan_sip.c: In function âadd_vcodec_to_sdpâ:
>> chan_sip.c:10077: warning: missing terminating " character
>> chan_sip.c:10077: error: missing terminating " character
>> chan_sip.c:10078: error: âprofileâ undeclared (first use in this function)
>> chan_sip.c:10078: error: (Each undeclared identifier is reported only once
>> chan_sip.c:10078: error: for each function it appears in.)
>> chan_sip.c:10078: error: âlevelâ undeclared (first use in this function)
>> chan_sip.c:10078: error: âidâ undeclared (first use in this function)
>> chan_sip.c:10078: error: invalid suffix "f" on integer constant
>> chan_sip.c:10078: error: expected â)â before â;â token
>> chan_sip.c:10079: error: stray â\â in program
>> chan_sip.c:10079: error: stray â\â in program
>> chan_sip.c:10079: warning: missing terminating " character
>> chan_sip.c:10079: error: missing terminating " character
>> chan_sip.c:10080: error: expected â;â before â}â token
>> make[1]: *** [chan_sip.o] Error 1
>> make: *** [channels] Error 2
> --
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