VoIP Mailing List Archives

[mikael at bjerkeland.com]

Hi,

I have been testing inbound calls to a Nokia phone with handover to a
cellphone number if I get MEDIA_TIMEOUT on the B leg of the call, and
had to set rtp-timeout to a very low 6 seconds in order to get "fast"
handover. This introduces an interesting side-effect that hangs up calls
even in the ringing state after 6 seconds. Is this the desired behaviour
of rtp-timeout-sec? My initial guess was that rtp-timeout-sec should
only be valid for established calls where the two endpoints have
exchanged rtp at some point but have stopped exchanging media. As far as
I know a phone call in ringing state has not shared any RTP with the
other endpoint until it gets early media or is answered. Should
rtp-timeout-sec really be valid even when ringing?

It seems to me that setting rtp-timeout-sec to 60 seconds would add an
absolute time limit on ringing phone calls to 60 seconds, which I
believe is not the actual purpose of this limit. Could anyone please
share their thoughts on this matter?


Thanks,
Mikael





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[anthony.minessale at g...]

Are you geting 183+sdp from the nokia?
the media timer only operates once media is established and only
counts against you if the channel is being read from and that does not
happen until you get a 183 or 200 w/sdp

try putting a debug line in switch_rtp.c around 1520
printf("MISSED PACKETS %u/%u\n", rtp_session->missed_count, rtp_session->max_missed_packets); 

but try updating first there was a recent fix that may have prevented a timer surge at the beginning of calls.


On Tue, Apr 28, 2009 at 6:20 AM, Mikael Aleksander Bjerkeland <mikael@bjerkeland.com (mikael@bjerkeland.com)> wrote:

[mikael at bjerkeland.com]

The scenario I was referring to was actually an outbound call from a
locally registered SIP phone to a cellphone. The same thing happens
whether I use a SIP or PRI trunk. After 6 s it hangs up.


I get SDP on 183 no matter whether I'm calling a cellphone or a fixed
line. I also get ringing indication. The 183+sdp is passed on to the
Nokia and after 6 s the call is hung up. Both the SIP and PRI trunks
claim to send early media but there seems to be no audio/RTP. If I
answer the call in 6 s it's not dropped because the media path was
established before RTP timeout.

The same thing happens on latest trunk.
I added the debug line at 1520 and did make && /etc/init.d/freeswitch
stop && make install && /etc/init.d/freeswitch start but the debug line
didn't show up anywhere in the CLI.

Is my upstream provider doing something wrong in sending early media in
these cases? Seems pretty odd. It can be avoided by setting a higher
rtp-timeout-sec but it will still be an absolute timeout on ringing.


A transcript of the log:

send 1293 bytes to udp/[1.1.1.1]:5060 at 13:55:56.451865:

------------------------------------------------------------------------
INVITE sip:21651019@domain.appsvrslip11.prigw.com SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK
Route: <sip:21651019@1.1.1.1>
Max-Forwards: 69
From: "someone" <sip:23695000@2.2.2.2>;tag=m2SepeSZ63e3g
To: <sip:21651019@domain.appsvrslip11.prigw.com>
Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
CSeq: 114345142 INVITE
Contact: <sip:mod_sofia@2.2.2.2:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13175M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 383
P-Asserted-Identity: "someone" <sip:23695000@2.2.2.2>

v=0
o=FreeSWITCH 3718974841365302606 4309079514688066219 IN IP4 2.2.2.2
s=FreeSWITCH
c=IN IP4 2.2.2.2
t=0 0
m=audio 52706 RTP/AVP 9 8 0 3 101 13
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
m=video 52752 RTP/AVP 99
a=rtpmap:99 H264/90000

------------------------------------------------------------------------
2009-04-28 15:55:56 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state()
Channel
sofia/external-eth1/21651019@domain.appsvrslip11.prigw.com;fs_path=sip:21651019@1.1.1.1 entering state [calling][0]
recv 305 bytes from udp/[1.1.1.1]:5060 at 13:55:56.482864:

------------------------------------------------------------------------
SIP/2.0 100 Trying
From: "someone"<sip:23695000@2.2.2.2>;tag=m2SepeSZ63e3g
To: <sip:21651019@domain.appsvrslip11.prigw.com>
Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
CSeq: 114345142 INVITE
Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK
Content-Length: 0


------------------------------------------------------------------------
recv 1035 bytes from udp/[1.1.1.1]:5060 at 13:55:58.296906:

------------------------------------------------------------------------
SIP/2.0 183 Session Progress
From: "someone"<sip:23695000@2.2.2.2>;tag=m2SepeSZ63e3g
To:
<sip:21651019@domain.appsvrslip11.prigw.com>;tag=20134330840200942815366
Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc
CSeq: 114345142 INVITE
Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK
content-type: application/sdp
contact: <sip:1.1.1.1:5060;nt_end_pt=YM0
+~K!-.f0vfc830~P68.cio~H9zwgW0VyisWTdcaM26c610Xbo1.nfS.5NQt3mO~~70!-.f0vft815;nt_server_host=1.1.1.1>
supported: 100rel
x-nt-party-id: -/
allow: ACK
allow: BYE
allow: CANCEL
allow: INVITE
allow: OPTIONS
allow: INFO
allow: SUBSCRIBE
allow: REFER
allow: NOTIFY
allow: PRACK
server: CS2000_NGSS/9.0
Content-Length: 300

v=0
o=IWSPM 573585738 573585738 IN IP4 84.20.97.100
s=-
e=unknown@invalid.net
t=0 0
m=audio 45954 RTP/AVP 8 0 18 101
c=IN IP4 84.20.97.100
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 99
c=IN IP4 2.2.2.2
a=rtpmap:99 H264/90000

------------------------------------------------------------------------
2009-04-28 15:55:58 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state()
Channel
sofia/external-eth1/21651019@domain.appsvrslip11.prigw.com;fs_path=sip:21651019@1.1.1.1 entering state [proceeding][183]
2009-04-28 15:55:58 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state()
Remote SDP:
v=0
o=IWSPM 573585738 573585738 IN IP4 84.20.97.100
s=-
e=unknown@invalid.net
t=0 0
m=audio 45954 RTP/AVP 8 0 18 101
c=IN IP4 84.20.97.100
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 0 RTP/AVP 99
c=IN IP4 2.2.2.2
a=rtpmap:99 H264/90000

2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000:20]/[G722:9:8000:20]
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:1912
sofia_glue_tech_set_codec() Set Codec
sofia/external-eth1/21651019@domain.appsvrslip11.prigw.com;fs_path=sip:21651019@1.1.1.1 PCMA/8000 20 ms 160 samples
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp()
Set 2833 dtmf payload to 101
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp()
AUDIO RTP
[sofia/external-eth1/21651019@domain.appsvrslip11.prigw.com;fs_path=sip:21651019@1.1.1.1] 2.2.2.2 port 52706 -> 84.20.97.100 port 45954 codec: 8 ms: 20
2009-04-28 15:55:58 [DEBUG] switch_rtp.c:906 switch_rtp_create()
Starting timer [soft] 160 bytes per 20ms
2009-04-28 15:55:58 [NOTICE] sofia_glue.c:2573 sofia_glue_tech_media()
Pre-Answer
sofia/external-eth1/21651019@domain.appsvrslip11.prigw.com;fs_path=sip:21651019@1.1.1.1!
2009-04-28 15:55:58 [DEBUG] switch_channel.c:1736
switch_channel_perform_mark_pre_answered() Send signal
sofia/internal/mikael-nokia@fs.voip.domain [BREAK]
2009-04-28 15:55:58 [DEBUG] switch_ivr_originate.c:1972
switch_ivr_originate() sofia/internal/mikael-nokia@fs.voip.domain
receive message [PROGRESS]
2009-04-28 15:55:58 [INFO] mod_sofia.c:1377 sofia_receive_message()
Asked to send early media by sofia/internal/mikael-nokia@fs.voip.domain
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000:20]/[G722:9:8000:20]
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2931 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:1912
sofia_glue_tech_set_codec() Set Codec
sofia/internal/mikael-nokia@fs.voip.domain PCMA/8000 20 ms 160 samples
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2891 sofia_glue_negotiate_sdp()
Set 2833 dtmf payload to 98
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp()
AUDIO RTP [sofia/internal/mikael-nokia@fs.voip.domain] 10.100.4.192 port
58072 -> 10.247.3.253 port 49152 codec: 8 ms: 20
2009-04-28 15:55:58 [DEBUG] switch_rtp.c:906 switch_rtp_create()
Starting timer [soft] 160 bytes per 20ms
2009-04-28 15:55:58 [DEBUG] sofia_glue.c:2325 sofia_glue_activate_rtp()
Set comfort noise payload to 13
2009-04-28 15:55:58 [NOTICE] sofia_glue.c:2573 sofia_glue_tech_media()
Pre-Answer sofia/internal/mikael-nokia@fs.voip.domain!
2009-04-28 15:55:58 [INFO] mod_sofia.c:1420 sofia_receive_message() Ring
SDP:
v=0
o=FreeSWITCH 1240868886 1240868887 IN IP4 10.100.4.192
s=FreeSWITCH
c=IN IP4 10.100.4.192
t=0 0
m=audio 58072 RTP/AVP 8 98 13
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv



El mar, 28-04-2009 a las 07:50 -0500, Anthony Minessale escribió:

[anthony.minessale at g...]

as soon as FS sees 183 it expects media.

if they send 183 and no media it will most certainly timeout

On Tue, Apr 28, 2009 at 9:33 AM, Mikael Aleksander Bjerkeland <mikael@bjerkeland.com (mikael@bjerkeland.com)> wrote:

[mikael at bjerkeland.com]

Thanks! I'll notify them of the problem and see if there's a way around it.



2009/4/28 Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)>