June 2013 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] freeswitch SBC register configure
by truong ngoc THANH 21 Jul '13

21 Jul '13

hi all, Does any one know how to using freeswitch as SBC server for register. Like, sip client send REGISTER to FS, FS forwarding this REGISTER message to kamailio, kamailio authen and return pass/false to FS, FS check if pass, store contact into local database, else send back to sip client. Please suggest to get this function work, Thanks, ThanhT

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[Freeswitch-dev] sofia profile stop response
by Xijing Dai 06 Jul '13

06 Jul '13

hey all, I have experienced a lot of times that sofia profile stops to response for any sip messages. And when you type sofia status, that profile is gone, but if we check the port by netstat, the port is still opened by freeswitch. This happens if we keep freeswitch running in 1 - 2 days. And there is no error log. Any idea on how to debug? Cheers.

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[Freeswitch-dev] Video input for FreeSwitch
by Thorhallur Sverrisson 05 Jul '13

05 Jul '13

Hi All, I've been working on setting up FreeSWITCH on my Raspberry PI as a replacement for my aging front door intercom. I have most of the quirks worked out using PortAudio for the audio and some shell script magic to handle the button input. As this is project is nearing its end I'ld like to add on the camera module for rpi and get streaming video from the front door. I know FS has h.264 passthrough, and I've seen some mp4 stuff showing up in the source tree although I have not dug too deep into that yet. I'm wondering if anyone has looked into this before, or if my idea is too wild to even consider. The rpi has a nice and simple command line program to interface to the camera, so 'all' that would be needed, if I understand correctly, is to execute the program with the right parameters and stream the standard output to the remote end. Any information or tips would be highly appreciated, Thor

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Re: [Freeswitch-dev] [Freeswitch-users] origin/master's HEAD is segfaulting
by Ken Rice 05 Jul '13

05 Jul '13

Oh come on people... Jira... That's where you report bugs... And yes a segfault is a bug On 6/22/13 9:03 AM, "arif hossain" <aftnix(a)live.com> wrote: > Hi, > > I'm on this commit : > arif@khost:~/src/freeswitch$ git show HEAD > commit f8122e39c51240b99bf07a767afe2f8a698f1486 > > Now its segfaulting on startup. > >> From generated core : > > Program terminated with signal 11, Segmentation fault. > #0 0x00007f245cc28df1 in su_casematch (s1=0x7f245cc7fc57 "", > s2=0x3000000030 <Address 0x3000000030 out of bounds>) at su_string.c:224 > 224 unsigned char a = *s1++, b = *s2++; > > Clearly s2 is wrongly initialized. > > The backtrace : > > (gdb) bt > #0 0x00007f245cc28df1 in su_casematch (s1=0x7f245cc7fc57 "", > s2=0x3000000030 <Address 0x3000000030 out of bounds>) at su_string.c:224 > #1 0x00007f245cc377ff in tport_server_addrinfo (mr=0x7f243c003c30, > canon=0x7f243c003bfe "[::1]", family=0, host=0x7f245c958670 "::1", > service=0x7f243c003c04 "5060", protocol=0x7f245cc7fc56 "*", > transports=0x7f245c9589c0, return_addrinfo=0x7f245c958608) at tport.c:1814 > #2 0x00007f245cc370a1 in tport_bind_server (mr=0x7f243c003c30, > tpn=0x7f245c958770, transports=0x7f245c9589c0, public=tport_type_local, > tags=0x7f245c958730) at tport.c:1652 > #3 0x00007f245cc36762 in tport_tbind (self=0x7f243c003c30, > tpn=0x7f245c958a50, transports=0x7f245c9589c0, > tag=0x7f245cecae20 <tptag_ident>, value=139794152182707) at tport.c:1524 > #4 0x00007f245cb8882d in nta_agent_add_tport (self=0x7f243c001e40, > uri=0x7f24400099c0, tag=0x7f245cecae20 <tptag_ident>, > value=139794152182707) at nta.c:2237 > #5 0x00007f245cbc4e71 in nua_stack_init_transport (nua=0x7f2440007970, > tags=0x7f24400098f0) at nua_register.c:1216 > #6 0x00007f245cbd6b6b in nua_stack_init (root=0x7f243c001130, > nua=0x7f2440007970) at nua_stack.c:194 > #7 0x00007f245cc24441 in su_pthread_port_clone_main (varg=0x7f245c9d03d0) > at su_pthread_port.c:340 > #8 0x00007f2460ffae9a in start_thread () > from /lib/x86_64-linux-gnu/libpthread.so.0 > ---Type <return> to continue, or q <return> to quit--- > #9 0x00007f2460d27ccd in clone () from /lib/x86_64-linux-gnu/libc.so.6 > #10 0x0000000000000000 in ?? () > > > In tport_server_addrinfo() : > > (gdb) list tport_server_addrinfo > ... > 1800 char const * const transports[], > ... > ... > 1811 for (i = 0, N = 0; transports[i] && N < TPORT_N; i++) { > 1812 su_addrinfo_t *ai = &hints[N]; > 1813 > 1814 if (!su_casematch(protocol, transports[i]) && > !su_strmatch(protocol, "*")) > > So char const * const transports[] is not correctly initialized before using > its values in the for loop. > > (gdb) p transports[0] > $3 = 0x3000000030 <Address 0x3000000030 out of bounds> > (gdb) p transports[1] > $4 = 0x7f245c958d00 "`\215\225\\$\177" > > So transports[0] is the problematic initialization here. > > Going back further, in libs/sofia-sip/libsofia-sip-ua/nta/nta.c: 2150 > > char const * const * tports = tports_sip; > > And here > > (gdb) ptype tports_sip > type = const char * const[5] > (gdb) p tports_sip[0] > $6 = 0x7f245cc7fc38 "udp" > (gdb) p tports[0] > $7 = 0x3000000030 <Address 0x3000000030 out of bounds> > (gdb) ptype tports_sip > type = const char * const[5] > (gdb) p tports > $11 = (const char * const *) 0x7f245c9589c0 > (gdb) p &tports_sip > $12 = (const char * const(*)[5]) 0x7f245cebf660 <tports_sip> > > As it can be seen, tports diverged from initial tports_sip. > It has been done in following snippet : > > 2197 if (strchr(tp, ',')) { > 2198 int i; char *t, *tps[9] = {0}; > 2199 > 2200 /* Split tp into transports */ > 2201 for (i = 0, t = tp; t && i < 8; i++) { > 2202 tps[i] = t; > 2203 if ((t = strchr(t, ','))) > 2204 *t++ = '\0'; > 2205 } > 2206 > 2207 tps[i] = NULL; > 2208 tports = (char const * const *)tps; > > > Now : > > (gdb) p tp > $20 = "udp\000tcp\000H\215\225\\$\177\000\000\240\252\354\\".. > > So above code should parse "udp" in tports[0]. Which is not the case here. > > I'm assuming this bug has to generated from a more higher level, rather than > happened in a file which has not been touched since 2006( I git blamed it). > > But bt does not have any calls to "non-sofia" stuff. > > This su_casematch() was called from two threads, > > 4 Thread 0x7f245eccb700 (LWP 7622) 0x00007f245cc28df1 in su_casematch ( > s1=0x7f245cc7fc57 "", s2=0x3000000030 <Address 0x3000000030 out of > bounds>) > at su_string.c:224 > > * 1 Thread 0x7f245c959700 (LWP 7623) 0x00007f245cc28df1 in su_casematch ( > s1=0x7f245cc7fc57 "", s2=0x3000000030 <Address 0x3000000030 out of > bounds>) > at su_string.c:224 > > > Both calls to su_casematch seems problematic. > > Sorry for a long BUGREPORT. Actually i thought i should post my investigation > if it helps the devs(After git blaming, i highly suspect that my investigation > unearthed nothing , problem generated from elsewhere, which i could not find). > > I've not started git bisecting. If it helps, i will be happy to do that. > > > > > > -- > Cheers > arif > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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[Freeswitch-dev] Wrong match between codecs offered by caller UAC and enabled codecs when multiple sample rates are in use
by Giacomo Vacca 05 Jul '13

05 Jul '13

Hi all, I've seen a behaviour similar to http://jira.freeswitch.org/browse/FS-5398and I have a proposed patch that fixes it. Scenario: - Caller offers, in order, AMR-WB, iLBC, SILK/12K, SILK/24K, SILK/8, speex/8 - FS has global_codec_prefs set to SILK@24000h@20i,SILK@16000h @20i,SILK@8000h@20i,speex@16000h@20i,speex@8000h@20i,PCMU,PC MA - FS decides to accept with SILK/16K (not even offered): the caller BYEs immediately FS logged: 7dc4b656-a7cf-46fe-93f0-b10b497a1f7d 2013-06-27 14:17:30.726808 [DEBUG] switch_core_media.c:2662 Audio Codec Compare [SILK:97:12000:20:0]/[SILK:120:24000:20:40000] 7dc4b656-a7cf-46fe-93f0-b10b497a1f7d 2013-06-27 14:17:30.726808 [DEBUG] switch_core_media.c:2662 Audio Codec Compare [SILK:97:12000:20:0]/[SILK:119:16000:20:30000] 7dc4b656-a7cf-46fe-93f0-b10b497a1f7d 2013-06-27 14:17:30.726808 [DEBUG] switch_core_media.c:1772 Set Codec sofia/internal/4656729(a)truphone.comSILK/16000 20 ms 320 samples 30000 bits To fix this I've added the comparison of sample rates in switch_core_media.c, with this patch (applied on latest master): diff --git a/src/switch_core_media.c b/src/switch_core_media.c index 9270b0b..4b926ed 100644 --- a/src/switch_core_media.c +++ b/src/switch_core_media.c @@ -2788,7 +2788,7 @@ SWITCH_DECLARE(uint8_t) switch_core_media_negotiate_sdp(switch_core_session_t *s if ((zstr(map->rm_encoding) || (smh->mparams->ndlb & SM_NDLB_ALLOW_BAD_IANANAME)) && map->rm_pt < 96) { match = (map->rm_pt == imp->ianacode) ? 1 : 0; } else { - match = strcasecmp(rm_encoding, imp->iananame) ? 0 : 1; + match = (!strcasecmp(rm_encoding, imp->iananame) && (map->rm_rate == codec_rate)) ? 1 : 0 } Hope this helps, even if a better solution exists. Regards, Giacomo

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[Freeswitch-dev] Error building RPM
by Worker Bee 05 Jul '13

05 Jul '13

Hello Everyone; I am new to FreeSwitch. I am attempting to create an rpm distribtion on CentOS6, Freeswitch version 1.2.9 I keep running into the following error: libtool: link: `/usr/lib64/libogg.la' is not a valid libtool archive I would be very appreciative if anyone could shed some light or point me in the right direction. Thanks! Bee

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[Freeswitch-dev] Github and FreeSWITCH
by Brian West 28 Jun '13

28 Jun '13

Please use http://jira.freeswitch.org to publish patches. -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest

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[Freeswitch-dev] help to correct sip stack to send to kamailio in mod_sofia
by ThanhTruong 28 Jun '13

28 Jun '13

hi all, I am very new in sofia sip, and I am develop one register module for FS using sofia sip stack. in FS, when i receive REGISTER message, i am trying to send it to another registrar server (kamailio). I have implement my code in incoming REGISTER message like: nua_set_params(profile->nua, SIPTAG_FROM_STR("sip:101@192.168.1.112:5062"), NUTAG_ENABLEMESSAGE(1), NUTAG_ENABLEINVITE(1), NUTAG_AUTOALERT(1), NUTAG_SESSION_TIMER(0), NUTAG_AUTOANSWER(0), TAG_NULL()); nua_register(nh, NUTAG_PROXY("sip:192.168.1.112:5062"), NUTAG_REGISTRAR("sip:192.168.1.112:5062"), //registra uri SIPTAG_FROM_STR("sip:192.168.1.112:5062"), NUTAG_INSTANCE (0), NUTAG_M_USERNAME("101"), TAG_NULL()); I have one account 101/xxxx in kamailio, i want to register from softphone to FS, FS send register and process authentication with kamailio, then return 200OK back to softphone. With this code bellow, i send REGISTER message and get response in kamailio as: send 613 bytes to udp/[192.168.1.112]:5062 at 03:10:42.833439: ------------------------------------------------------------------------ REGISTER sip:101@192.168.3.122:42806;rinstance=7742cc70900998e8 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.112;rport;branch=z9hG4bKBB44UFQ9ySjQK Max-Forwards: 70 From: <sip:192.168.1.112:5062>;tag=DXeN4j2DFt27j To: "101" <sip:101@192.168.1.112>;tag=cd424d7d Call-ID: ac980e34-4e79-1231-71ba-3e508d3991e2 CSeq: 45210457 REGISTER Contact: <sip:101@192.168.1.112> User-Agent: FreeSWITCH-mod_sofia/1.5.2b+git~20130527T075833Z~f3541e00f4 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 recv 325 bytes from udp/[192.168.1.112]:5062 at 03:10:42.834220: ------------------------------------------------------------------------ SIP/2.0 404 Not here Via: SIP/2.0/UDP 192.168.1.112;rport=5060;branch=z9hG4bKBB44UFQ9ySjQK From: <sip:192.168.1.112:5062>;tag=DXeN4j2DFt27j To: "101" <sip:101@192.168.1.112>;tag=cd424d7d Call-ID: ac980e34-4e79-1231-71ba-3e508d3991e2 CSeq: 45210457 REGISTER Server: kamailio (4.0.1 (x86_64/linux)) Content-Length: 0 Then, I see that structure of SIP REGISTER message is not correct. Can some one help me to construct this message to send to kamailio ? May we dont care about nua_create ? just using nua_set_params or nua_register or nua_options ? Thank all in advance. ThanhTruong

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[Freeswitch-dev] ReINVITE failure with ACK+SDP scenario
by BNML 28 Jun '13

28 Jun '13

Hi all, I am having trouble with a transfert failure in a special scenario which i wasn't able to find documentation or configuration option solving it. I have a PBX creating two outbound call, it want to bridge them on the FreeSWITCH. It is sending a first INVITE without SDP and it will send the SDP in the ACK. (This is not breaking RFC3261 but I admit rarely seen scenario. http://tools.ietf.org/html/rfc3261#section-13.2.1 : "The UAC MAY choose to add a message body to the INVITE.". Looking at this scenario there is, indeed, at least one interesting scenario where it can helps avoiding creating another useless Re-INVITE transaction, see below) For the second call it is sending a re-invite with SDP. FreeSWITCH will understand and forward (because of "bypass_media") the second Re-INVITE but does never forward the first Re-INVITE (The one without SDP in the INVITE but in the ACK) to carrier, so this will end up in one-way-audio. A little schema to help : PBX FREESWITCH COMMENT 1) INVITE (a-leg) [THERE IS NO SDP] 2) 200 OK (fs-a-leg)[FreeSWITCH answer with the SDP information before any media update occurs, but my PBX is now aware of media informations] 3) INVITE+SDP (b-leg) [This is b-leg updated media information with previous 200 OK (fs-a-leg)] 4) 200 OK (fs-b-leg)[My PBX updated media and has b-leg freeswitch media information that it will send in the ACK+SDP to finalize the both-legs Re-INVITE] 5) ACK (b-leg) [Acknoledgement of b-leg Re-INVITE transaction] 6) ACK+SDP (a-leg) [ *** problem *** it assume FreeSWITCH will update media information and forward to the carrier which it won't] The solution i imagine is FreeSWITCH SHOULD trigger a Re-INVITE either for the a-leg at INVITE (without SDP in "1") receiption (and update media information when receiving the ACK+SDP) _OR_ create a Re-INVITE transaction at "6" when receiving ACK+SDP. One way or another, I am supposing FreeSWITCH refuse to forward the INVITE without SDP at first maybe because of carrier compatibility ? If not, this seems the way to go. Has anyone faced such a scenario ? Am I rigth assuming a patch is needed ? If yes, any suggestions are welcome. Thanks you BNML

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[Freeswitch-dev] Multiple ptime in SDP
by Wolfgang Pichler 28 Jun '13

28 Jun '13

Hi all, i did have a problem while doing a interconnection to a customer, with multiple ptime in sdp. His equipment did not accepted the sdp freeswitch offered *m=audio 52628 RTP/AVP 0 8 3 9 18 101 13* *a=rtpmap:101 telephone-event/8000* *a=fmtp:101 0-16* *a=ptime:20* *m=audio 52628 RTP/AVP 4 101 13* *a=rtpmap:101 telephone-event/8000* *a=fmtp:101 0-16* *a=ptime:30* * * *to avoid this - i have added sdp_m_per_ptime=false before bridge. * * * *I have done some readings about this - and it is not clear to me if freeswitch accts according to this draft http://tools.ietf.org/html/draft-garcia-mmusic-multiple-ptimes-problem-01* *It seems to be a mix between method 2 and method 6 - but i am not sure.* * * *Can anyone clearify which method freeswitch is using here ?* * * *br,* *Wolfgang Pichler*

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