Re: [Sofia-sip-devel] Custom Headers in 200 OK to UPDATE requests Freeswitch
On 02.09.24 11:59, Souradeep Bera wrote: Hello, I need to pass some custom headers from Freeswitch in the 200 OK response sent for an incoming UPDATE request. But I think the sofia sip stack is automatically sending the response, not even allowing me to add any custom headers from mod_sofia. Is there anyway I can achieve this? My SIP flow is something like Freeswitch Remote Trunk INVITE -> <-- 100 (INVITE) <-- 183 (INVITE) <-- UPDATE 200 (UPDATE). --> <-- 200 (INVITE) ACK > I require custom headers in the 200 OK sent from Freeswitch to the remote trunk for the update request Hey there, I think you'd have to ask on the Freeswitch mailing list/forum whether that is possible from within Freeswitch. This list is about the SIP stack only, though it's pretty dead nowadays. Best regards, Andreas ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Reading custom SIP headers
On 08/29/2016 08:32 AM, John Nash wrote:
> I am looking for some way to read value of a custom SIP header (starting
> with X-). I have access to object ssip_t. Which function I can use to
> extract custom SIP header and use its value?
>
>
>
> --
>
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The sip structure contains a member of 'sip_unknown'.
You can use it like this to iterate custom (or "unknown") SIP headers:
sip_unknown_t* unknown_header = sip->sip_unknown;
while( unknown_header )
{
if( 0 == strcmp( unknown_header->un_name, "X-MyCustomHeader" ) )
{
/* whatever... */
}
unknown_header = unknown_header->un_next;
}
Regards
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[Sofia-sip-devel] [FIX] crash when querying adapter addresses failed (Windows)
During long and heavy testruns I encountered seemingly random crashes running on Windows 7. I tracked the problem down to the function static int win_localinfo(su_localinfo_t const hints[1], su_localinfo_t **rresult); which resides in su_localinfo.c. If the call to GetAdapterAddresses() fails, the function tries to use FormatMessage() to get an error string to print out. Unfortunately, the call to FormatMessage() specifies an illegal buffer: if (!FormatMessage(FORMAT_MESSAGE_ALLOCATE_BUFFER | FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_IGNORE_INSERTS, NULL, error, MAKELANGID(LANG_NEUTRAL, SUBLANG_DEFAULT), msg, 0, NULL)) msg = empty; It SHOULD be: if (!FormatMessage(FORMAT_MESSAGE_ALLOCATE_BUFFER | FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_IGNORE_INSERTS, NULL, error, MAKELANGID(LANG_NEUTRAL, SUBLANG_DEFAULT), (LPTSTR)&msg, 0, NULL)) msg = empty; Earlier in the function it says: char const* empty = ""; LPTSTR msg = empty; Calling FormatMessage() and specifying a pointer to empty (assignment) is obviously illegal as it requires the address to a pointer. Best regards, Andreas -- Site24x7 APM Insight: Get Deep Visibility into Application Performance APM + Mobile APM + RUM: Monitor 3 App instances at just $35/Month Monitor end-to-end web transactions and take corrective actions now Troubleshoot faster and improve end-user experience. Signup Now! http://pubads.g.doubleclick.net/gampad/clk?id=272487151&iu=/4140 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] nua_unsubscribe
I've never used the subscription mechanism before, but you might try to
use NUTAG_M_USERNAME(x).
Regards,
Andreas
On 06/01/2015 11:41 PM, Johannes Bauer wrote:
> Hi everybody!
>
> Fisrt I want to thank you developers for your excelent work on this SIP
> framework, But as you sure can guess, I have a problem.
> I am using sofia sip to get notified when a busy callee is ready again, using
> a SUBSCRIBE for a dialog event.
> The first SUBSCRIBE works perfectly, I get a NOTIFY "confirmed" for
> subscribing and the NOTIFY "terminate" from the callee, when the current call
> has ended. However when I want to unsubscribe i get error 404 from the callee.
> It seems that the user of the SUBSCRIBE Request is missing. It should be
> "sip:test@192.168.90.100" but only "sip:192.168.90.100" is sent on the
> SUBSCRIBE with expire 0.
>
> code looks like this:
>
> case nua_i_notify:
>
>
>
> printf("Notify received, status: %d, %s\n", status, phrase);
>
> state_start = 0;
> state_end = 0;
> state_start = strstr(sip->sip_payload->pl_data, "");
> state_end = strstr(sip->sip_payload->pl_data, "");
>
> if((state_start) && (state_end))
> {
> state_start += 7; // start after
> strncpy_s(state, sizeof(state), state_start, state_end -
> state_start);
> printf("state changed to %s\n",state);
>
> if (!strcmp(state, "terminated"))
>
> {
> nua_unsubscribe(nh, SIPTAG_EVENT_STR("dialog"),
> TAG_END());
>
>
> Is it necessary to make a new nua_handle for the unsubscribe.
> When is it necessary to genereate a new handle for operations?
>
> Kind regards
>
>
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[Sofia-sip-devel] nua_handle_has_register() not working
Hi! I discovered, that nua_handle_has_register() always returns 0 even when the handle has been used with nua_register()/nua_unregister(). From what I can see in the code, the same goes for nua_handle_has_subscribe() but not for nua_handle_has_invite(). It looks as if the call to nua_stack_set_handle_special() in the client transaction has simply been forgotten? The FreeSwitch fork seems to have the same issue. Regards, Andreas -- Meet PCI DSS 3.0 Compliance Requirements with EventLog Analyzer Achieve PCI DSS 3.0 Compliant Status with Out-of-the-box PCI DSS Reports Are you Audit-Ready for PCI DSS 3.0 Compliance? Download White paper Comply to PCI DSS 3.0 Requirement 10 and 11.5 with EventLog Analyzer http://pubads.g.doubleclick.net/gampad/clk?id=154622311&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] Session timer stops when receiving '491 Request pending'
Subject says it all. I ran into an issue with a buggy Asterisk where the re-INVITE (refresh) from my app would overlap with a re-INVITE from the Asterisk. Both parties respond with '491 Request pending' and that's it. The bug in Asterisk is that it never actually retries the re-INVITE (yes, it's a very old version; no, an upgrade is _not_ possible *grrr*). The effect is, that I've got a dangling call (re-INVITE from Asterisk came because the caller hung up) because my application only hangs up calls if some kind of higher level error occurs. When I understand RFC3261 correctly, the UAC must retry the request after some time and since (again: if I understand correctly) the UAC/UAS roles are transaction based: Shouldn't BOTH parties retry their INVITE requests? I'll have to backport the fix in Asterisk but isn't this a problem in Sofia, too? Regards, Andreas -- Meet PCI DSS 3.0 Compliance Requirements with EventLog Analyzer Achieve PCI DSS 3.0 Compliant Status with Out-of-the-box PCI DSS Reports Are you Audit-Ready for PCI DSS 3.0 Compliance? Download White paper Comply to PCI DSS 3.0 Requirement 10 and 11.5 with EventLog Analyzer http://pubads.g.doubleclick.net/gampad/clk?id=154622311&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] Request specific handle?
Hi!
Is there a way to somehow provide an outgoing request with a
context/magic specific to that request?
In my case, the user application shall be able to send SIP INFO requests
inside an active call.
The application provides a "request_status" callback in which the final
response of the request is to be delivered.
I have to assume that the application wants to send a couple of SIP INFO
requests shortly after one another.
I've got a nua handle magic, but that one is for the entire call.
I was thinking of something like this:
struct info_ctxt
{
my_callback_t cb;
};
void sendInfo( my_callback_t cb, )
{
info_ctxt = (info_ctxt*)malloc( sizeof( info_ctxt ) );
info_ctxt->cb = cb;
nua_info( nua_handle,
NUTAG_RMAGIC_REF( info_ctxt ),
.,
TAG_END() );
}
void sofia_callback( )
{
switch( event )
{
...
case nua_r_info:
{
info_ctxt* ctxt = NULL;
tl_gets( tags, NUTAG_RMAGIC_REF( ctxt ), TAG_END() )
}
...
}
}
Am I dreaming, or do I have to implement my own queue and send INFO
requests after getting nua_r_info
when a preceding request has completed?
Best regards,
Andreas
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Re: [Sofia-sip-devel] How to change media mode in active call?
On 09/04/2014 03:19 PM, Michael Jerris wrote: > You can do the actual sdp structure as well. For sure you will need the m > lines in the sdp, if you do the structure, it can be sparse just like the > string. > > I rewrote my function to create the SDP as string and supply it to NUA using USER_SDP_STR, that works just fine. Another question though: I tried to use SOATAG_LOCAL_SDP and SOATAG_REMOTE_SDP within my nua_i_state callback, but it gives me an assertion: WARNING: tag soa::remote_sdp directly used by tl_gets() Assertion failed: tt->tt_class == ref_tag_class Is it illegal to use these tags in that context? Their _REF version are working. How would I get the structure to the local SDP when I'm not in the nua_i_state callback? Would I use soa_get_local_sdp() using the session object directly from the nua handle? I mean like so: soa_get_local_sdp( nua_handle->nh_soa, &l_sdp, NULL, NULL ); I don't feel safe accessing members of the nua handle directly, that's why I'm asking. Thanks a lot, Andreas -- Slashdot TV. Video for Nerds. Stuff that matters. http://tv.slashdot.org/ ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] Typo in soa_tag.h
Hi! I just came across a problem that prevents the use of SOATAG_LOCAL_SDP_STR_REF. In soa_tag.h this SOFIAPUBVAR tag_typedef_t soatag_session_sdp_str_ref; should be SOFIAPUBVAR tag_typedef_t soatag_local_sdp_str_ref; MSVC complained about the undeclared identifier 'soatag_local_sdp_str_ref' when I used the tag with tl_gets while handling nua_i_state. The current Freeswitch master seems to have the problem also. Regards, Andreas -- Slashdot TV. Video for Nerds. Stuff that matters. http://tv.slashdot.org/ ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] How to change media mode in active call?
Alright, from the beginning: --- That's how I create the stack: sip_stack = nua_create( sip_root, sip_evt_dispatcher, reinterpret_cast< nua_magic_t* >( this ), TAG_IF( (! uas_id.empty()), SIPTAG_USER_AGENT_STR( uas_id.c_str() ) ), SIPTAG_ALLOW_STR( "INVITE,ACK,CANCEL,BYE,OPTIONS,PRACK,UPDATE,MESSAGE" ), NUTAG_URL( my_url.c_str() ), TAG_END() ); --- Placing a call involves these calls: nh = nua_handle( sip_stack, reinterpret_cast< nua_hmagic_t* >( this ), SIPTAG_TO( to ), TAG_IF( (NULL != from), SIPTAG_FROM( from ) ), TAG_IF( ((NULL == from) && (strlen(from_user ? from_user : "") > 0)), NUTAG_M_USERNAME( from_user ) ), TAG_IF( session_timer.useSessionTimer(), NUTAG_SESSION_REFRESHER( nua_local_refresher ) ), TAG_IF( session_timer.useSessionTimer(), NUTAG_MIN_SE( session_timer.minimumSessionExpires() ) ), TAG_IF( session_timer.useSessionTimer(), NUTAG_SESSION_TIMER( session_timer.sessionExpires() ) ), TAG_IF( session_timer.useSessionTimer(), NUTAG_REFRESH_WITHOUT_SDP( 1 ) ), TAG_END() ); nua_invite( nh, TAG_IF( (! additional_header_str.empty()), SIPTAG_HEADER_STR( additional_header_str.c_str() ) ), SOATAG_USER_SDP_STR( sdp_os.str().c_str() ), SOATAG_RTP_SELECT( SOA_RTP_SELECT_COMMON ), TAG_END() ); --- Accepting a call looks like this: nua_respond( reinterpret_cast< nua_handle_t* >( call_id() ), status_code, sip_status_phrase( status_code ), TAG_IF( strlen( sdp_str ), SOATAG_USER_SDP_STR( sdp_str ) ), TAG_IF( (! aux_codecs.empty()), SOATAG_AUDIO_AUX( aux_codecs.c_str() ) ), TAG_IF( strlen( reason_str ), SIPTAG_REASON_STR( reason_str ) ), TAG_IF( use_session_timer, NUTAG_SESSION_REFRESHER( nua_local_refresher ) ), TAG_IF( use_session_timer, NUTAG_MIN_SE( session_timer.minimumSessionExpires() ) ), TAG_IF( use_session_timer, NUTAG_SESSION_TIMER( session_timer.sessionExpires() ) ), TAG_IF( use_session_timer, NUTAG_REFRESH_WITHOUT_SDP( 1 ) ), TAG_END() ); --- My first (naive) attempt was trying to update the media mode like so: std::string sdp_media_mode_line( "a=" ); sdp_media_mode_line.append( media_mode.sdp_str() ); nua_invite( reinterpret_cast< nua_handle_t* >( call_id() ), SOATAG_USER_SDP_STR( sdp_media_mode_line.c_str() ), TAG_END() ); Of course this doesn't work since SOA doesn't know what media that attribute belongs to... I didn't try to include the media line (yet). What I'm basically trying to do is to change a certain attribute for all media of a certain type at once. Is there a more elegant way than USER_SDP_STR, or is it the recommended way to go? Regards, Andreas 03.09.2014, 15:23, "Michael Jerris" : > I know it works with USER_SDP_STR, what exactly is your call to > update/invite? are you doing ACTIVE_AUDIO(1) ? > > On Sep 3, 2014, at 9:13 AM, Andreas Wehrmann wrote: >> Hi! >> >> I'm somewhat at a loss here. >> I'm in a SIP call that I made or accepted with one of >> a=[inactive|recvonly|sendonly|sendrecv] in the SDP. >> There is only one media (audio). >> When I make or answer a call, I create the complete SDP and provide it >> to nua with SOATAG_USER_SDP_STR which is working just fine. >> >> Now I would like to be able to change the media mode mid-call using the >> same SDP as before with a new media mode setting, but how do I do that? >> I tried calling nua_update/nua_invite using SOATAG_ACTIVE_AUDIO but it >> was to no avail. >> >> Is SOATAG_ACTIVE_AUDIO the way to go or should I be doing something >> different? >> >> Any help is appreciated. >> >> Best regards, >> Andreas -- Slashdot TV. Video for Nerds. Stuff that matters. http://tv.slashdot.org/ ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] How to change media mode in active call?
On 09/03/2014 03:13 PM, Andreas Wehrmann wrote: > Hi! > > I'm somewhat at a loss here. > I'm in a SIP call that I made or accepted with one of > a=[inactive|recvonly|sendonly|sendrecv] in the SDP. > There is only one media (audio). > When I make or answer a call, I create the complete SDP and provide it > to nua with SOATAG_USER_SDP_STR which is working just fine. > > Now I would like to be able to change the media mode mid-call using the > same SDP as before with a new media mode setting, but how do I do that? > I tried calling nua_update/nua_invite using SOATAG_ACTIVE_AUDIO but it > was to no avail. > > Is SOATAG_ACTIVE_AUDIO the way to go or should I be doing something > different? > > Any help is appreciated. > > Best regards, > Andreas > > > > I forgot to mention, that the SDP sent out when calling nua_update/nua_invite is not even the original SDP but a new minimal SDP that doesn't seem to reflect the current state of the call/media. -- Slashdot TV. Video for Nerds. Stuff that matters. http://tv.slashdot.org/ ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] How to change media mode in active call?
Hi! I'm somewhat at a loss here. I'm in a SIP call that I made or accepted with one of a=[inactive|recvonly|sendonly|sendrecv] in the SDP. There is only one media (audio). When I make or answer a call, I create the complete SDP and provide it to nua with SOATAG_USER_SDP_STR which is working just fine. Now I would like to be able to change the media mode mid-call using the same SDP as before with a new media mode setting, but how do I do that? I tried calling nua_update/nua_invite using SOATAG_ACTIVE_AUDIO but it was to no avail. Is SOATAG_ACTIVE_AUDIO the way to go or should I be doing something different? Any help is appreciated. Best regards, Andreas -- Slashdot TV. Video for Nerds. Stuff that matters. http://tv.slashdot.org/ ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] any way to bind client to specific interface(eth/wlan).
On 08/02/2014 02:08 AM, Luca Olivetti wrote: > El 02/08/14 00:28, Nikos Balkanas ha escrit: > >> Oh, and the other patches are needed too >> >> http://code.google.com/p/danube-voip/source/browse/sofia-sip/patches/ >> >> Bye >> -- >> Luca >> >> >> Any reason why these were never commited to main branch? > No idea, but (and I could be wrong) I suppose the original developers > have abandoned the project, given what happened with nokia. > > Does anyone of the FreeSwitch guys have commit-access to the Sofia repository? Maybe with their help we could update and improve Sofia's main branch. -- Want fast and easy access to all the code in your enterprise? Index and search up to 200,000 lines of code with a free copy of Black Duck Code Sight - the same software that powers the world's largest code search on Ohloh, the Black Duck Open Hub! Try it now. http://p.sf.net/sfu/bds ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] any way to bind client to specific interface(eth/wlan).
On 08/01/2014 08:17 AM, Saiyam Doshi wrote:
> Hello,
>
> In nua_create(), providing NUATAG_URL("sip:*:*") is taking 127.0.0.1 as
> client iP, if i call nua_create prior to any interface available(Ethernet an
> Wlan both are down).
>
> After this even if i up ethernet and it gets ip, all request are routed via
> 127.0.0.1 ip only.
>
> I have checked NUTAG_DETECT_NETWORK_UPDATES(), but its only works with OS x.
> I am using linux.
>
>
> So, is there any way to bind client to take ip from specific
> interface(eth/wlan).
>
> Or any tag can be used with nua_create() to do the same.
>
> Thanks.
>
> Regards,
> Saiyam Doshi
>
Hi!
I took a look at the code and it looks like this is only implemented for
OSx.
Regards,
Andreas
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Re: [Sofia-sip-devel] Example compilation
Well, the complete SDP as a char string you want the stack to respond with. If you don't know what SDP is, you should probably read a little bit more about SIP in general: SIP: http://www.rfc-editor.org/rfc/rfc3261.txt SDP: http://www.rfc-editor.org/rfc/rfc2327.txt I don't have too much time to look into this, but I think there are a lot of test programs that compile when you build Sofia. Maybe you should start there, I believe there is a simple UA client among them... Regards, Andreas On 07/21/2014 08:19 AM, Nikos Balkanas wrote: > Ty. > > What does it expect for an SDP string? What does it mean? What is it used > for? > > Seems these examples are outdated. I managed to fix some of them (operation > for op_t, etc.) but I still have some more i could use some help: > > nua_callstate_t (undeclared identifier) and NUTAG__REF. > > I also assume that oper_ctx_[u,t] is the same as operation. These examples > have lots of errors... > > > On Mon, Jul 21, 2014 at 8:16 AM, Andreas Wehrmann > wrote: > >> On 07/21/2014 05:06 AM, Nikos Balkanas wrote: >>> I'm going through the examples in the site. >>> >>> In app_i_invite it states: >>> >>>nua_respond(nh, 200, "OK", SOA_USER_SDP(magic->sdp), TAG_END()); >>> >>> Looking at the nua_respond documentation, SOA_USER_SDP is a tag, that is >>> not defined and gcc errs. Furthermore, magic is my application and >> doesn't >>> have an sdp field. >>> These seem like hard errors. What can i do? >>> >>> TIA, >>> Nikos >>> >> Hi, >> >> SOA_USER_SDP should indeed be SOATAG_USER_SDP which comes from here: >> libsofia-sip-ua/soa/sofia-sip/soa_tag.h >> >> You can build your own SDP string and provide it using SOATAG_USER_SDP_STR. >> I guess that's how most people use it. >> >> Regards, >> Andreas >> >> >> >> -- >> Want fast and easy access to all the code in your enterprise? Index and >> search up to 200,000 lines of code with a free copy of Black Duck >> Code Sight - the same software that powers the world's largest code >> search on Ohloh, the Black Duck Open Hub! Try it now. >> http://p.sf.net/sfu/bds >> ___ >> Sofia-sip-devel mailing list >> Sofia-sip-devel@lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel >> -- Want fast and easy access to all the code in your enterprise? Index and search up to 200,000 lines of code with a free copy of Black Duck Code Sight - the same software that powers the world's largest code search on Ohloh, the Black Duck Open Hub! Try it now. http://p.sf.net/sfu/bds ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Example compilation
On 07/21/2014 05:06 AM, Nikos Balkanas wrote: > I'm going through the examples in the site. > > In app_i_invite it states: > > nua_respond(nh, 200, "OK", SOA_USER_SDP(magic->sdp), TAG_END()); > > Looking at the nua_respond documentation, SOA_USER_SDP is a tag, that is > not defined and gcc errs. Furthermore, magic is my application and doesn't > have an sdp field. > These seem like hard errors. What can i do? > > TIA, > Nikos > Hi, SOA_USER_SDP should indeed be SOATAG_USER_SDP which comes from here: libsofia-sip-ua/soa/sofia-sip/soa_tag.h You can build your own SDP string and provide it using SOATAG_USER_SDP_STR. I guess that's how most people use it. Regards, Andreas -- Want fast and easy access to all the code in your enterprise? Index and search up to 200,000 lines of code with a free copy of Black Duck Code Sight - the same software that powers the world's largest code search on Ohloh, the Black Duck Open Hub! Try it now. http://p.sf.net/sfu/bds ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Crash when system time jumps
On 06/12/2014 02:30 PM, Michael Jerris wrote:
> + /* check it there are still things queued, if there are, that means
> everything scheduled is > 15 days in the future */
> +/* in this case, we had a large time shift, we should schedule for 15
> days in the future (which is probably still before now) */
> + /* and this should sort itself out on the next run through */
> +if ( !agent->sa_out.completed->q_head && !agent->sa_out.trying->q_head
> && !agent->sa_out.inv_calling->q_head &&
> +!agent->sa_out.re_list &&
> !agent->sa_in.inv_confirmed->q_head && !agent->sa_in.preliminary->q_head &&
> +!agent->sa_in.completed->q_head &&
> !agent->sa_in.inv_completed->q_head && !agent->sa_in.re_list ) {
> SU_DEBUG_9(("nta: timer not set\n" VA_NONE));
Thanks for the patch.
I wasn't even aware of this issue.
Regards,
Andreas
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Re: [Sofia-sip-devel] ACK(SDP) is not send by UDP
Could you provide an example code to reproduce the problem? I cannot promise to find a solution, but I would like to take a look at the code. Regards, Andreas On 02/19/2014 08:05 PM, Guilherme César Pilon wrote: > Hello, > > I'm using sofia-sip-1.12.11 and i'm having o problem when i need to send an > ACK with a SDP and this message is bigger than 1400 > > bytes. The usual behavior would be to try sending the message by TCP (but > the calle does not accept TCP, only UDP), and if the > > connection is not established, send it by UDP anyway. But, when sofia tries > with TCP and the connection is not established, it does > > not send by UDP. > > Here is the log where the error occurs: > > nua: nua_stack_set_params: entering > nta: selecting scheme sip > tport_tsend(0x814cc10) tpn = */10.10.35.11:5060 > tport_resolve addrinfo = 10.10.35.11:5060 > tport_by_addrinfo(0x814cc10): not found by name */10.10.35.11:5060 > nta: ACK (900450185) too large for UDP, trying TCP > tport_tsend(0x814ced8) tpn = tcp/10.10.35.11:5060 > tport_resolve addrinfo = 10.10.35.11:5060 > tport_by_addrinfo(0x814ced8): not found by name tcp/10.10.35.11:5060 > tport_alloc_secondary(0x814ced8): new secondary tport 0x81538b0 > tport_base_connect(0x81538b0): connecting to tcp/10.10.35.11:5060/sip > tport(0x81538b0): set timer at 2000 ms because idle since send > tport_queue(0x81538b0): queueing 0x814f7a0 for tcp/10.10.35.11:5060 > nta: sent ACK (900450185) to tcp/10.10.35.11:5060 > nua(0x814fe58): call state changed: completing -> ready, sent answer > nua: nua_application_event: entering > nua: nua_application_event: entering > tport_connected(0x81538b0): events ERR > nta_agent: tport: Connection refused > tport_close(0x81538b0): tcp/10.10.35.11:5060/sip > tport_connected error1:0 > tport_connected error2:0 > nta: timer set next to 4495 ms > tport_wakeup_pri(0x814cc10): events IN > tport_recv_event(0x814cc10) > tport_recv_iovec(0x814cc10) msg 0x8157560 from (udp/10.10.107.204:5060) has > 347 bytes, veclen = 1 > > Just making it clear, i do not want the ACK sent by TCP, i want it sent by > UDP. Other messages bigger than ~1,4k are sent by UDP, just the ACK > > seems to have this problem. > > If someone could help me i would be grateful. > > > > -- > Managing the Performance of Cloud-Based Applications > Take advantage of what the Cloud has to offer - Avoid Common Pitfalls. > Read the Whitepaper. > http://pubads.g.doubleclick.net/gampad/clk?id=121054471&iu=/4140/ostg.clktrk > > > ___ > Sofia-sip-devel mailing list > Sofia-sip-devel@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel -- Flow-based real-time traffic analytics software. Cisco certified tool. Monitor traffic, SLAs, QoS, Medianet, WAAS etc. with NetFlow Analyzer Customize your own dashboards, set traffic alerts and generate reports. Network behavioral analysis & security monitoring. All-in-one tool. http://pubads.g.doubleclick.net/gampad/clk?id=126839071&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] nua_invite to IP Adress (DNS problem)
Before trying this and that and assuming something goes through, here's a suggestion: On your MAC: start Wireshark (you do know that, right?) and repeat the test. That way, you'll know exactly what goes where from whom and you'll be able to inspect the packets one by one. Wireshark actually has a nice feature that can list all VoIP calls it happened to notice while sniffing network traffic. Regards, Andy On 11/27/2013 07:25 PM, Alexander Heinrich wrote: > Actually the port should be open. > > Now I’m getting some new logs and a new experience. (I increased the > NTA_TIMER_C() to 1) > > When I try to call in my network I’m still getting a bad output: > > * tport_wakeup_pri(0x14d48680): events IN* > *tport_recv_event(0x14d48680)* > *tport_recv_iovec(0x14d48680) msg 0x14d1ab70 from (udp/192.168.178.145:5060) > has > 304 bytes, veclen = 1* > *tport_deliver(0x14d48680): msg 0x14d1ab70 (288 bytes) from > udp/192.168.178.82:5060/sip next=0x0* > *nta: received 100 Trying for INVITE (898641285)* > *nta: 100 Trying is going to a transaction* > *nta_outgoing: RTT is 420.385 ms* > *tport_release(0x14d48680): 0x14e8bc90 by 0x14d1a900 with 0x14d1ab70 > (preliminary)* > *nua: nua_application_event: entering* > ** > * * > * Event nua_r_invite * > * Status 100 * > * Trying* > * * > ** > * nta: timer set next to 9919 ms* > *nta: timer C fired, CANCEL and timeout INVITE (898641285)* > *tport_tsend(0x14d48680) tpn = UDP/192.168.178.82:5060* > *tport_resolve addrinfo = 192.168.178.82:5060* > *tport_by_addrinfo(0x14d48680): not found by name UDP/192.168.178.82:5060* > *tport_vsend returned 296* > *nta: sent CANCEL (898641285) to UDP/192.168.178.82:5060* > *tport_pend(0x14d48680): pending 0x14d1b190 for udp/192.168.178.145:5060 > (already 1)* > *nta_outgoing_timer: 0/0 resent, 1/0 tout, 0/0 term, 0/1 free* > *nta: timer set next to 500 ms* > *tport_wakeup_pri(0x14d48680): events IN* > *tport_recv_event(0x14d48680)* > *tport_recv_iovec(0x14d48680) msg 0x14e8b470 from (udp/192.168.178.145:5060) > has > 833 bytes, veclen = 1* > *tport_deliver(0x14d48680): msg 0x14e8b470 (399 bytes) from > udp/192.168.178.82:5060/sip next=0x0* > *nta: received 200 OK for CANCEL (898641285)* > *nta: 200 OK is going to a transaction* > *nta_outgoing: RTT is 240.489 ms* > *tport_release(0x14d48680): 0x14d1b190 by 0x14d1b6c0 with 0x14e8b470* > *tport_wakeup_pri(0x14d48680): events IN* > *tport_recv_event(0x14d48680)* > *tport_recv_iovec(0x14d48680) msg 0x14e8adb0 from (udp/192.168.178.145:5060) > has > 418 bytes, veclen = 1* > *tport_deliver(0x14d48680): msg 0x14e8adb0 (402 bytes) from > udp/192.168.178.82:5060/sip next=0x0* > *nta: received 487 Request Terminated for INVITE (898641285)* > *nta: 487 Request Terminated is going to a transaction* > *tport_release(0x14d48680): 0x14e8bc90 by 0x14d1a900 with 0x14e8adb0* > *tport_tsend(0x14d48680) tpn = UDP/192.168.178.82:5060* > *tport_resolve addrinfo = 192.168.178.82:5060* > *tport_by_addrinfo(0x14d48680): not found by name UDP/192.168.178.82:5060* > *tport_vsend returned 290* > *nta: sent ACK (898641285) to UDP/192.168.178.82:5060* > *nta: outgoing_free(0x14d1b8e0)* > *nua(0x14e8c170): call state changed: calling -> init* > *nua(0x14e8c170): removing session usage* > *soa_destroy(static::0x14d1d780) called* > *nta_leg_destroy(0x14e8bb80)* > *nua: nua_application_event: entering* > ** > * * > * Event nua_r_invite * > * Status 487 * > * Request Terminated* > * * > > > But in the recent calls section of my Mac App I can see that someone was > trying > to call. > As an attachment I’m sending you a screenshot. > > I tried the TPORT_Log thing (seen in the output) and I think RTP packets > should > come through, because calling from the Mac App to my iPhone works perfectly. > Greetings, > Alexander Heinrich > > > On 27 Nov 2013, at 10:58, Dave Horton <mailto:d...@dchorton.com>> wrote: > > > Are you actually listening for RTP packets on the port you are advertising > > in > > the INVITE (2) ? If not, and the SIP signaling you are generating > > otherwise looks ok, I would suspect that the Mac app experiences failure in > > sending initial RTP to that port (receives ICMP destination unreachable > > since > > nothing is listening on that port) and reacts according to what you are > > seeing. > > > > > > On Nov 27, 2013, at 9:09 AM, Alexander Heinrich > <mailto:alexan...@sn0wfreeze.de>> wrote: > > > > I thought that, too > > > > So I downloaded a SIP App from the Appstore and tried to call the Mac App > > with > > the SIP Ap
Re: [Sofia-sip-devel] nua_invite to IP Adress (DNS problem)
But then I guess it's a problem on the other end since there is nothing else
you can do.
If the Mac-App responds with 100 - Trying it MUST have received your INVITE
(otherwise where would 100 come from?).
Did you write the Mac-App yourself or is it some third-party app you're trying
to call?
I would suggest that if you're trying to write a Soft-phone that you get your
hands on another
Soft-phone which is known to work. So you have something to test your
application against.
Regards,
Andreas
Alexander Heinrich schrieb am 9:13 Mittwoch,
27.November 2013:
Yes this is actually the problem.
When I’m trying to call somebody the App which should receive the call doesn’t
not start to ring and I’m not able to accept the call.
But I know, that it receiving a message from the caller, because the Mac App
starts to bounce in the dock like it should do.
I hope you understand what I mean.
It works like that
iPhone —> Sending invite on local Network —> Mac - App
iPhone <— 100 Trying <— Mac-App
And that’s it. The Mac-App does not show, that it gets an invite. And it’s
actually not ringing.
Thanks for your assistance.
Greetings,
Alexander Heinrich
On 27 Nov 2013, at 09:01, Andreas Wehrmann wrote:
Actually, this looks ok to me.
>You send an INVITE and get a response of 100 - Trying which is perfectly
>normal.
>Whether the peer you're calling sends 180 - Ringing or not is up to him since
>this is optional.
>
>The log "not found by name" should not worry you, I'm not entirely sure what
>Sofia is trying to do there,
>I'm guessing it's trying to do a reverse lookup which probably fails resulting
>in this supposed error log
>(I get the same log in my application but everything works).
>
>What problem remains now?
>Can you not establish the call?
>
>Regards,
>Andreas
>
>
>
>
>
>Alexander Heinrich schrieb am 22:21 Dienstag,
>26.November 2013:
>
>Thanks for your help, but this still doesn’t work for me.
>
>
>Here’s my output:
>
>
> nta: timer A fired, retransmit INVITE (89860)
>tport_release(0x14deafc0): 0x14ef5bd0 by 0x14ef6330 with 0x0
>tport_tsend(0x14deafc0) tpn = */192.168.178.82:5060
>tport_resolve addrinfo = 192.168.178.82:5060
>tport_by_addrinfo(0x14deafc0): not found by name */192.168.178.82:5060
>tport_vsend returned 687
>nta: resent INVITE (89860) to */192.168.178.82:5060
>
>
>And that’s my actual code:
>
>
> url_string_t *url = URL_STRING_MAKE(sipAddress) ;
> sip_to_t *to = sip_to_create(NULL, url);
>
>
> op.handle = nua_handle(app.nua, NULL,SIPTAG_TO(to), TAG_END());
>
>nua_invite(op.handle,SOATAG_USER_SDP_STR("m=audio 2 RTP/AVP 0\n"
> "a=rtpmap:0 PCMU/8000\n"
> "v = 0\n"
> "o=Alex\n"
> "s=Audio"),
> SIPTAG_TO(to),
> TAG_END());
>
>I don’t know why the address could not be found.
>tport_by_addrinfo(0x14deafc0): not found by name */192.168.178.82:5060
>
>
>Some of you said, that binding to a specifc interface could help. So I tried
>binding to my IP:
> app.nua= nua_create(app.root,
> event_callback,
> NULL,
> NUTAG_URL("sip:192.168.178.145:5060"),
> NUTAG_AUTOALERT(1),
> NTATAG_PASS_100(1),
> NTATAG_PASS_408(1),
> TAG_END());
>And I’m actually getting a different output.
>
>
>I’ll try to explain what the output says, so you don’t have to read everything:
>I’m sending an invite to the correct IP address. The SIP client is answering
>100 Trying like it should do.
>Normally that should be followed by 180 ringing automatically.
>But at the end nta says, that the timer is not set.
>
>
>I tried setting different timers, too. Is there a specific tag for the nta
>timer? So it could start ringing automatically ?
>
>
>
>
>
>
>
> Event nua_i_state
> Status 0
> INVITE sent
>
>
> nta: timer A fired, retransmit INVITE (898622277)
>tport_release(0x15557090): 0x1554ea60 by 0x1554cd10 with 0x0
>tport_tsend(0x15557090) tpn = */192.168.178.82:5060
>tport_resolve addrinfo = 192.168.178.82:5060
>tport_by_addrinfo(0x15557090): not found by name */192.168.178.82:5060
>tport_vsend returned 697
>nta: resent INVITE (898622277) to */192.168.178.82:5060
>tport_pend(0x15557090): pending 0x1554ea60 for udp/192.168.17
Re: [Sofia-sip-devel] nua_invite to IP Adress (DNS problem)
Actually, this looks ok to me.
You send an INVITE and get a response of 100 - Trying which is perfectly normal.
Whether the peer you're calling sends 180 - Ringing or not is up to him since
this is optional.
The log "not found by name" should not worry you, I'm not entirely sure what
Sofia is trying to do there,
I'm guessing it's trying to do a reverse lookup which probably fails resulting
in this supposed error log
(I get the same log in my application but everything works).
What problem remains now?
Can you not establish the call?
Regards,
Andreas
Alexander Heinrich schrieb am 22:21 Dienstag,
26.November 2013:
Thanks for your help, but this still doesn’t work for me.
Here’s my output:
nta: timer A fired, retransmit INVITE (89860)
tport_release(0x14deafc0): 0x14ef5bd0 by 0x14ef6330 with 0x0
tport_tsend(0x14deafc0) tpn = */192.168.178.82:5060
tport_resolve addrinfo = 192.168.178.82:5060
tport_by_addrinfo(0x14deafc0): not found by name */192.168.178.82:5060
tport_vsend returned 687
nta: resent INVITE (89860) to */192.168.178.82:5060
And that’s my actual code:
url_string_t *url = URL_STRING_MAKE(sipAddress) ;
sip_to_t *to = sip_to_create(NULL, url);
op.handle = nua_handle(app.nua, NULL,SIPTAG_TO(to), TAG_END());
nua_invite(op.handle,SOATAG_USER_SDP_STR("m=audio 2 RTP/AVP 0\n"
"a=rtpmap:0 PCMU/8000\n"
"v = 0\n"
"o=Alex\n"
"s=Audio"),
SIPTAG_TO(to),
TAG_END());
I don’t know why the address could not be found. tport_by_addrinfo(0x14deafc0):
not found by name */192.168.178.82:5060
Some of you said, that binding to a specifc interface could help. So I tried
binding to my IP:
app.nua= nua_create(app.root,
event_callback,
NULL,
NUTAG_URL("sip:192.168.178.145:5060"),
NUTAG_AUTOALERT(1),
NTATAG_PASS_100(1),
NTATAG_PASS_408(1),
TAG_END());
And I’m actually getting a different output.
I’ll try to explain what the output says, so you don’t have to read everything:
I’m sending an invite to the correct IP address. The SIP client is answering
100 Trying like it should do.
Normally that should be followed by 180 ringing automatically.
But at the end nta says, that the timer is not set.
I tried setting different timers, too. Is there a specific tag for the nta
timer? So it could start ringing automatically ?
Event nua_i_state
Status 0
INVITE sent
nta: timer A fired, retransmit INVITE (898622277)
tport_release(0x15557090): 0x1554ea60 by 0x1554cd10 with 0x0
tport_tsend(0x15557090) tpn = */192.168.178.82:5060
tport_resolve addrinfo = 192.168.178.82:5060
tport_by_addrinfo(0x15557090): not found by name */192.168.178.82:5060
tport_vsend returned 697
nta: resent INVITE (898622277) to */192.168.178.82:5060
tport_pend(0x15557090): pending 0x1554ea60 for udp/192.168.178.145:5060
(already 0)
nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
nta: timer set next to 1000 ms
tport_wakeup_pri(0x15557090): events IN
tport_recv_event(0x15557090)
tport_recv_iovec(0x15557090) msg 0x1554b1c0 from (udp/192.168.178.145:5060) has
304 bytes, veclen = 1
tport_deliver(0x15557090): msg 0x1554b1c0 (288 bytes) from
udp/192.168.178.82:5060/sip next=0x0
nta: received 100 Trying for INVITE (898622277)
nta: 100 Trying is going to a transaction
nta_outgoing: RTT is 510.482 ms
tport_release(0x15557090): 0x1554ea60 by 0x1554cd10 with 0x1554b1c0
(preliminary)
nua: nua_application_event: entering
Event nua_r_invite
Status 100
Trying
tport_wakeup_pri(0x15557090): events IN
tport_recv_event(0x15557090)
tport_recv_iovec(0x15557090) msg 0x156b52c0 from (udp/192.168.178.145:5060) has
304 bytes, veclen = 1
tport_deliver(0x15557090): msg 0x156b52c0 (288 bytes) from
udp/192.168.178.82:5060/sip next=0x0
nta: received 100 Trying for INVITE (898622277)
nta: 100 Trying is going to a transaction
tport_release(0x15557090): 0x1554ea60 by 0x1554cd10 with 0x156b52c0
(preliminary)
nua: nua_application_event: entering
Event nua_r_invite
Status 100
Trying
nta: timer not set
Greetings,
Alexander Heinrich --
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Re: [Sofia-sip-devel] nua_invite to IP Adress (DNS problem)
On 11/25/2013 12:20 AM, Alexander Heinrich wrote:
> Good evening dear developers,
>
> I’m currently stuck in another problem.
> I am able to retrieve incoming calls. So my nua_create should be right.
>
> In my nua_invite I’m trying to connect to an IP Address directly. I am using
> IP4, but the computer which I try to call does run IP6, too.
> When I try to connect sofia sip is not able to find out the way to my
> computer and I get a timeout in the end.
>
> I am still using an iOS Device for testing, so I am not able to use the DNS
> resolver.
> Is it possible to deactivate the resolver directly in my code ?
>
> Here’s my invite:
>
>url_string_t *url = URL_STRING_MAKE(“sip:alex@192.168.178.82:5060") ;
> sip_to_t *to = sip_to_create(NULL, url);
>
> op.handle = nua_handle(app.nua, NULL,SIPTAG_TO(to), TAG_END());
> //op.leg = nta_leg_tcreate(app.agent,nta_callback, NULL, SIPTAG_TO(to),
> TAG_END());
>
> nua_invite(op.handle,SOATAG_USER_SDP_STR("m=audio 2 RTP/AVP 0\n"
> "a=rtpmap:0 PCMU/8000\n"
> "v = 0\n"
> "o=Alex\n"
> "c=IN IP4 127.0.0.1\n"
> "s=Audio"),
> TAG_END());
> Here’s my Log Output:
>
> --- cut ---
>
>
> When I try using the SOATAG_ADDRESS Feature:
>
>nua_invite(op.handle,SOATAG_USER_SDP_STR("m=audio 2 RTP/AVP 0\n"
> "a=rtpmap:0 PCMU/8000\n"
> "v = 0\n"
> "o=Alex\n"
> "c=IN IP4 127.0.0.1\n"
> "s=Audio"),
> SOATAG_ADDRESS("192.168.178.82")
> TAG_END());
>
>
>
You seem to be confusing addresses in the SIP header and media addresses.
With c= or SOATAG_ADDRESS you put an address in the SDP you're sending out;
that is: The address you want the remote peer to send media to (will
most probably be the host you're calling from).
The other thing is the SIP URI you're trying to call which is a
completely different story.
Creating a TO header works like this:
sip_to_t* to = sip_to_create( NULL, URL_STRING_MAKE(
"sip:alex@192.168.178.82:5060" ) );
nua_invite( handle, SIPTAG_TO( to ), ..., TAG_END() );
Hope this helps,
Andreas
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[Sofia-sip-devel] Fwd: Re: Sending nua_invite - DNS Error 503
Original Message Subject:Re: [Sofia-sip-devel] Sending nua_invite - DNS Error 503 Date: Tue, 19 Nov 2013 21:19:37 +0100 From: Alexander Heinrich To: andreas_wehrm...@yahoo.at Hey Andreas, you were right but how can I disable the DNS Lookup in Sofia Sip? Do you have any idea ? Greetings, Alexander Heinrich I wouldn't know; In my projects I don't use names at all, just naked addresses. Did you try Tim Na's Patch? Regards -- Shape the Mobile Experience: Free Subscription Software experts and developers: Be at the forefront of tech innovation. Intel(R) Software Adrenaline delivers strategic insight and game-changing conversations that shape the rapidly evolving mobile landscape. Sign up now. http://pubads.g.doubleclick.net/gampad/clk?id=63431311&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] status of sofia sip
On 11/18/2013 02:15 PM, Michael Jerris wrote: > Pekka is no longer at Nokia. I think I am the only one with commit access to > the gitorious tree still around and responding at all. I don't have the > ability to roll releases. There are still plenty of people working on > sofia-sip from what I can see, but yeah, no official maintainer who is around > and actively working on anything. > > Mike > > So am I right to assume that Pekka doesn't participate anymore and there is no lead maintainer at all? Andreas -- DreamFactory - Open Source REST & JSON Services for HTML5 & Native Apps OAuth, Users, Roles, SQL, NoSQL, BLOB Storage and External API Access Free app hosting. Or install the open source package on any LAMP server. Sign up and see examples for AngularJS, jQuery, Sencha Touch and Native! http://pubads.g.doubleclick.net/gampad/clk?id=63469471&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] status of sofia sip
On 11/18/2013 10:04 AM, Dave Horton wrote: > Hi Andreas - > > Yes, I went back and read your question. Looks like it never really got > answered, and then sidetracked into a specific bug fix discussion. And while > it would be great to get fixes from Freeswitch back in, that clearly isn't > happening yet. If I am reading the commit history in gitorious correctly (as > you'd mentioned, just finding out that is where it was hosted was a fairly > random occurrence), things ground to a halt a couple of years ago with > Pekka's last commit. Also, Freeswitch uses nua but I prefer to use nta in > most of my projects. I see a handful of merge requests being opened but not > being addressed. > > For now I've created my own clone on gitorious, like a few others have done. > I suppose like others that have done this I will can cherry-pick commits that > look useful from other people's clones, but this is slapdash and does kind of > both concern me and sadden me to see a useful product that seems to be dying > slowly from lack of support. I'd be interested in feedback that either (a) > tells me I'm wrong, and why or (b) provides suggestion on other (modern, > supported, open, C/C++) sip stacks that I ought to be looking at. > > Dave > > I would need to agree with you there. I was often wondering who is responsible for Sofia. That would seem to be Pekka but he seems to be concerned with other things (I'm not blaming him, just pointing it out) and there's apparently nobody else? What I find a little bit odd is that the official branch of Sofia seems to have come to a halt although there seem to be lots of people using it. The question that bothers me is: Why didn't changes from these people flow back to the official version? (guess that's because Pekka is (the only?) maintainer and just didn't have the time). The FreeSwitch guys seem to do a good job but the I think the sofia-sip version in the FreeSwitch repo is to be considered a fork, not just a branch of the official repo? I think this library needs more maintainers. Thoughts, anyone? -- DreamFactory - Open Source REST & JSON Services for HTML5 & Native Apps OAuth, Users, Roles, SQL, NoSQL, BLOB Storage and External API Access Free app hosting. Or install the open source package on any LAMP server. Sign up and see examples for AngularJS, jQuery, Sencha Touch and Native! http://pubads.g.doubleclick.net/gampad/clk?id=63469471&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] status of sofia sip
On 11/17/2013 04:53 PM, Dave Horton wrote: > I've been using sofia sip quite happily for a few years in various projects, > but I am wondering about whether it is still being actively supported. I'm > heading into a new project where I will need a lot of 3GPP headers that don't > appear to have been implemented in sofia sip, and I've also come across a bug > or two in the current header implementation. I'd be happy to chip in with > patches, but I'm not clear whether this code base is being actively supported > by any key person or group I should coordinate with. > Hey there! I've posted the same questions a couple of weeks ago. Sofia-SIP is being actively maintained in the FreeSWITCH project (see http://www.freeswitch.org/). If you take a look at their GIT history (http://fisheye.freeswitch.org/browse/freeswitch.git; under libs/sofia-sip) you will find quite a few changes since the last official release. I would like to see those changes merged back too, but to me it seems this is not going to happen soon. Regards, Andreas -- DreamFactory - Open Source REST & JSON Services for HTML5 & Native Apps OAuth, Users, Roles, SQL, NoSQL, BLOB Storage and External API Access Free app hosting. Or install the open source package on any LAMP server. Sign up and see examples for AngularJS, jQuery, Sencha Touch and Native! http://pubads.g.doubleclick.net/gampad/clk?id=63469471&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Sending nua_invite - DNS Error 503
On 11/18/2013 12:12 AM, Huang, Kun-Yao wrote: > Try binding to a specific interface instead of 0.0.0.0. Sofia SIP chooses an > arbitrary interface for out-going requests when you bind to 0.0.0.0 > > http://www.mail-archive.com/sofia-sip-devel@lists.sourceforge.net/msg04427.html > http://www.mail-archive.com/sofia-sip-devel@lists.sourceforge.net/msg03185.html > > Kun-Yao Huang > > I don't have any experience with iOS, but how about resolving the DNS manually and then use the resolved address with Sofia? I know this is ugly... Regards, Andreas -- DreamFactory - Open Source REST & JSON Services for HTML5 & Native Apps OAuth, Users, Roles, SQL, NoSQL, BLOB Storage and External API Access Free app hosting. Or install the open source package on any LAMP server. Sign up and see examples for AngularJS, jQuery, Sencha Touch and Native! http://pubads.g.doubleclick.net/gampad/clk?id=63469471&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Starting with Sofia SIP(registrating)
Your NUTAG_URL is flawed, it must represent the local-URL where your device can
be reached, try "0.0.0.0"
Regards,
Andreas
Alexander Heinrich schrieb am 0:46 Dienstag,
12.November 2013:
Hey developers,
I am currently working on an iPhone port of Sofia SIP.
I already got the library compiled for the iPhone and its running.
But now I am not getting any further.
I used the short message example provided by gitorious for sending and
receiving messages.
And now I am struggling here:
* Connecting to a SIP Server ( in this case iptel.org)
Here’s my code. I don’t know what is right and what isn’t
nua = nua_create(root,
event_callback,
NULL,
NUTAG_REGISTRAR("sip:sfdsip:vghl9...@iptel.org:5060"),
SIPTAG_FROM_STR("sip:sfdsip:vghl9...@iptel.org:5060"),
NUTAG_M_USERNAME("sfdsip"),
NUTAG_URL("sip:sfdsip:vghl9...@iptel.org:5060"),
TAG_END());
My Output:
su_socket_port_init(0x1565f460, 0x10e19c) called
su_pthread_port_init(0x1565f460, 0x10e19c) called
nua: nua_create: entering
su_socket_port_init(0x15562270, 0x10e19c) called
su_pthread_port_init(0x15562270, 0x10e19c) called
nua: nua_stack_init: entering
nua: nua_stack_set_params: entering
soa_create("default", 0x15582e40, 0x155a1e20) called
soa_set_params(static::0x15679800, ...) called
soa_set_params(static::0x15679800, ...) called
nta_agent_create: initialized hash tables
nta_agent_create: initialized transports
nta_agent_create: initialized random identifiers
nta_agent_create: initialized timer
nta_agent_create: initialized resolver
tport_create(): 0x15664fc0
nta: master transport created
tport_bind_server(0x15664fc0) to */iptel.org:5060/sip
tport_bind_server(0x15664fc0): calling tport_listen for udp
tport_alloc_primary(0x15664fc0): new primary tport 0x155a1560
tport_listen(0x15664fc0): bind(pf=2 udp/[212.79.111.155]:5060): Can't assign
requested address
tport_bind_server(0x15664fc0): calling tport_listen for tcp
tport_alloc_primary(0x15664fc0): new primary tport 0x155a1560
tport_listen(0x15664fc0): bind(pf=2 tcp/[212.79.111.155]:5060): Can't assign
requested address
nta: bind(iptel.org:5060;transport=*): Can't assign requested address
nua: initializing SIP stack failed
nua: nua_stack_deinit: entering
sres_sofia_update(0x15661230, -1, -1)
sres_sofia_update(0x0, -1, -1)
tport_destroy(0x15664fc0)
su_kqueue_port_deinit(0x15562270) called
I tried the nua_register method, too.
I don’t know which one I have to use for my registration at iptel.org.
I want that every SIP call works over my SIP Account at iptel
(Very much I’s by the way… Perhaps I am a bit egoistic 😉)
I hope there are still some cool developers, who could help me here.
Because this topic is really not that public.
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Re: [Sofia-sip-devel] Cant using nua_invite to initial call.
On 08/09/2013 11:21 AM, ThanhTruong wrote: > Hi , > > this is full loglevel for all and set to 9(max) I think. > > in this stack, i can receive invite, send/receive register, respond of > register. > > :( > But with invite, it is not live nua_register, nua_subscribe so, i am getting > this issue. > > Thank you for your help. > > > ThanhTruong > > > > Maybe you should write a little test app in which you do nothing but initiate an outgoing call. If that works, you can try to integrate that into your existing app and try to find the differences if it doesn't work. Regards, Andreas -- Get 100% visibility into Java/.NET code with AppDynamics Lite! It's a free troubleshooting tool designed for production. Get down to code-level detail for bottlenecks, with <2% overhead. Download for free and get started troubleshooting in minutes. http://pubads.g.doubleclick.net/gampad/clk?id=48897031&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Cant using nua_invite to initial call.
On 08/09/2013 10:57 AM, ThanhTruong wrote: > Hi sir, > > When I tried with your code, change to my sdp and get this log from > nua_invite. > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:282 soa_clone() soa_clone(static::0xe9bd050, 0xe490080, 0x2b52284121d0) > called > soa.c:405 soa_set_params() soa_set_params(static::0x2b521c27d9d0, ...) called > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > soa.c:405 soa_set_params() soa_set_params(static::0x2b521c27d9d0, ...) called > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua_dialog.c:338 nua_dialog_usage_add() nua(0x2b52284121d0): adding session > usage > nta.c:4322 nta_leg_tcreate() nta_leg_tcreate(0x2b521860a290) > soa.c:1302 soa_init_offer_answer() > soa_init_offer_answer(static::0x2b521c27d9d0) called > soa.c:1426 soa_generate_offer() soa_generate_offer(static::0x2b521c27d9d0, 0) > called > nua_stack.c:271 nua_stack_event() nua(0x2b52284121d0): event r_invite 900 > Internal error at nua_client.c:711 > nua_session.c:4135 signal_call_state_change() nua(0x2b52284121d0): call state > changed: init -> init > nua_stack.c:271 nua_stack_event() nua(0x2b52284121d0): event i_state 900 > Internal error at nua_client.c:711 > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > nua_stack.c:271 nua_stack_event() nua(0x2b52284121d0): event i_terminated 900 > Internal error at nua_client.c:711 > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x2b52284121d0): removing > session usage > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > nua_stack.c:359 nua_application_event() nua: nua_application_event: entering > soa.c:358 soa_destroy() soa_destroy(static::0x2b521c27d9d0) called > nta.c:4375 nta_leg_destroy() nta_leg_destroy(0x2b521860a290) > > Let me tell more about my scenario: > + first I have one nh to handle incoming INVITE from my client. > + when I get one invite from client(xlite) I call your code to send one > invite to another sip server. and get this error. > code is: > nua_handle_t* nh_a = NULL; > sip_to_t* to = sip_to_create( NULL, URL_STRING_MAKE( > "sip:372@59.120.208.196" ) ); > nh_a = nua_handle( nua, NULL, >SIPTAG_TO( to ), >TAG_END() ); > nua_invite( nh_a, > SOATAG_USER_SDP_STR( r_sdp), > TAG_END() ); > > and I see we get 900 Internal error. > :( Is it possible to increase the log level even more? I don't see why this shouldn't work; maybe it has to do something with how you setup your SIP stack in the first place? But that's just a wild guess. -- Get 100% visibility into Java/.NET code with AppDynamics Lite! It's a free troubleshooting tool designed for production. Get down to code-level detail for bottlenecks, with <2% overhead. Download for free and get started troubleshooting in minutes. http://pubads.g.doubleclick.net/gampad/clk?id=48897031&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Cant using nua_invite to initial call.
On 08/09/2013 10:30 AM, ThanhTruong wrote: > Hi, > > I have used ngrep -p -q -W byline port 5060 in this server and did not see > any invite message sent out. > > I think sofia-sip handling it and drop it, not send it out. > > is there any way to get it work ? do you have any example code to send invite > to one sip server ? (only how to use nua_invite to send one INVITE message to > one server) > > Thanks, > > > ThanhTruong > > > ____ > From: Andreas Wehrmann > To: ThanhTruong > Cc: "sofia-sip-devel@lists.sourceforge.net" > > Sent: Friday, August 9, 2013 3:19 PM > Subject: Re: [Sofia-sip-devel] Cant using nua_invite to initial call. > > > > This is a (very much) stripped down example from my code (NOTE: that I didn't include the SDP string here!): nua_handle_t* nh = NULL; sip_to_t* to = sip_to_create( NULL, URL_STRING_MAKE( "sip:bla@something" ) ); nh = nua_handle( sip_stack, NULL, SIPTAG_TO( to ), TAG_END() ); nua_invite( nh, SOATAG_USER_SDP_STR( sdp_os.str().c_str() ), TAG_END() ); -- Get 100% visibility into Java/.NET code with AppDynamics Lite! It's a free troubleshooting tool designed for production. Get down to code-level detail for bottlenecks, with <2% overhead. Download for free and get started troubleshooting in minutes. http://pubads.g.doubleclick.net/gampad/clk?id=48897031&iu=/4140/ostg.clktrk ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Cant using nua_invite to initial call.
On 08/09/2013 09:48 AM, ThanhTruong wrote:
> hi,
>
> I have enabled log level and see some log but cant know where is my issue ?
>
> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
> nua.c:633 nua_invite() nua: nua_invite: entering
> nua_stack.c:529 nua_signal() nua(0x2b52184f5940): sent signal r_invite
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua_stack.c:569 nua_stack_signal() nua(0x2b52184f5940): recv signal r_invite
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:405 soa_set_params() soa_set_params(static::0x2b5218638ba0, ...) called
> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x2b5218638ba0, (nil),
> 0x2b521c32bca4, -1) called
> soa.c:890 soa_set_capability_sdp()
> soa_set_capability_sdp(static::0x2b5218638ba0, (nil), 0x2b521c32bca4, -1)
> called
> soa.c:2568 soa_init_sdp_connection_with_session()
> soa_init_sdp_connection_with_session: selected IN IP4 (a local
> address)
> nua_stack.c:271 nua_stack_event() nua(0x2b52184f5940): event r_invite 491
> Request Pending
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>
> but Invite does not send out.
> nua_invite (nh, NUTAG_URL (nua_url),
> SOATAG_USER_SDP_STR("m=audio 8000 RTP/AVP 8 0\nc=IN IP4
> 127.0.0.1\n"),
> SOATAG_RTP_SORT(SOA_RTP_SORT_REMOTE),
> SOATAG_RTP_SELECT(SOA_RTP_SELECT_ALL),
> TAG_END());
>
>
This line looks suspicious:
nua_stack.c:271 nua_stack_event() nua(0x2b52184f5940): event r_invite 491
Request Pending
How did you make you make sure 'nothing goes out'?
Did you check with Wireshark?
Looking at the log, I would say your INVITE gets out but is rejected with 491.
Regards
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Re: [Sofia-sip-devel] Cant using nua_invite to initial call.
On 08/09/2013 06:32 AM, ThanhTruong wrote:
> Hi all,
>
> I am back and have a stupid question that i cant send invite to SIP server
> using nua_invite.
>
> i have tried :
>
> nua_invite (nh, NUTAG_URL (nua_url),
> SOATAG_ADDRESS("127.0.0.1:8000"),
> NUTAG_EARLY_MEDIA (1), TAG_END ());
>
> but I did not see any invite sent.
>
> any one can help to send one invite to one sip server ?
>
> thank all in advance.
>
>
> ThanhTruong
>
Increase the log level and find out what Sofia says about your request,
maybe that will give you a hint.
Regards,
Andreas
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Re: [Sofia-sip-devel] sip register message construct
On 06/13/2013 10:09 AM, Andreas Wehrmann wrote:
> Hey there!
>
> Nothing to do with the problem but I just wanted to say that you can
> simplify/strengthen
> the code for building the auth string by doing it like this code:
>
> "sip" is of type , which is delivered by NUA to the
> callback:
>
> sip_www_authenticate_t const* www_auth = sip->sip_www_authenticate;
> char const* scheme = www_auth->au_scheme;
> const char* realm = msg_params_find( www_auth->au_params, "realm=" );
> char auth_str[ 8192 ] = { 0 };
> snprintf( auth_str, 8192, "%s:%s:%s:%s",
> scheme, realm,
> "USERNAME",
> "PASSWORD" );
> nua_authenticate( nh, NUTAG_AUTH( auth_str ), TAG_END() );
>
>
Just to be thorough: The code above should check whether sheme or realm
is NULL;
Just in case.
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Re: [Sofia-sip-devel] sip register message construct
On 06/13/2013 09:34 AM, ThanhTruong wrote:
>
> For handling register authentication in callback I used:
>
> if(evt == nua_r_register && (status == 401 || status == 407))
>{
> if (sip && sip->sip_www_authenticate &&
> sip->sip_www_authenticate->au_params)
> {
> const char realm[] = "realm=\"";
> for (const char *param = (const
> char*)sip->sip_www_authenticate->au_params[0]; param != NULL; ++param)
> {
>size_t param_strlen = strlen(param);
>if (param_strlen > sizeof(realm) &&
> strncmp(realm, param, sizeof(realm) - 1) == 0)
>{
> std::string realm_val(param +
> sizeof(realm) - 1, param_strlen - sizeof(realm));
>std::stringstream digest;
> digest << "Digest:\"" << realm_val
> << "\"" << ":" << m_userName << ":" << m_passWord;
>
> nua_authenticate(m_registrationHandle, NUTAG_AUTH(digest.str().c_str()),
> TAG_END());
>}
> }
> }
>}
>
>
Hey there!
Nothing to do with the problem but I just wanted to say that you can
simplify/strengthen
the code for building the auth string by doing it like this code:
"sip" is of type , which is delivered by NUA to the
callback:
sip_www_authenticate_t const* www_auth = sip->sip_www_authenticate;
char const* scheme = www_auth->au_scheme;
const char* realm = msg_params_find( www_auth->au_params, "realm=" );
char auth_str[ 8192 ] = { 0 };
snprintf( auth_str, 8192, "%s:%s:%s:%s",
scheme, realm,
"USERNAME",
"PASSWORD" );
nua_authenticate( nh, NUTAG_AUTH( auth_str ), TAG_END() );
Hope it helps :-)
Regards,
Andreas
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Re: [Sofia-sip-devel] How to get default SIP_FROM host
On 09/03/2012 08:19 AM, Andreas Wehrmann wrote: > Hello everbody! > > If you create an outgoing request and don't provide a SIP FROM user, > sofia inserts one for you. > > Now, in my case I want to make an outgoing call and provide just the > user part (the calling number) and let the host part be chosen by sofia. > > Is there some way to do this? > Or is there a way to query the default host sofia would use? > > I tried to create a FROM user using the URI calling-number@0.0.0.0 > hoping that sofia would replace the dummy address with the real address > (like it does in SDP processing) but it wouldn't work. > > > Regards, > > Andreas > The public cannot see the example URI, so here it is again: calling-number (at) 0 . 0 . 0 . 0 Regards, Andreas -- Dipl.-Ing. (FH) Andreas Wehrmann Software Development -- Center Communication Systems GmbH A-1210 Wien, Ignaz-Köck-Straße 19 Sitz in Wien FN 796 88p, Firmenbuchgericht Wien www.centersystems.com Tel.: +43 (0) 190 199 - 3616 Mobile: +43 (0) 664 884 75916 Fax: +43 (0) 190 199 - 2110 E-Mail: a.wehrm...@centersystems.com -- Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] How to get default SIP_FROM host
Hello everbody! If you create an outgoing request and don't provide a SIP FROM user, sofia inserts one for you. Now, in my case I want to make an outgoing call and provide just the user part (the calling number) and let the host part be chosen by sofia. Is there some way to do this? Or is there a way to query the default host sofia would use? I tried to create a FROM user using the URI calling-number@0.0.0.0 hoping that sofia would replace the dummy address with the real address (like it does in SDP processing) but it wouldn't work. Regards, Andreas -- Dipl.-Ing. (FH) Andreas Wehrmann Software Development -- Center Communication Systems GmbH A-1210 Wien, Ignaz-Köck-Straße 19 Sitz in Wien FN 796 88p, Firmenbuchgericht Wien www.centersystems.com Tel.: +43 (0) 190 199 - 3616 Mobile: +43 (0) 664 884 75916 Fax: +43 (0) 190 199 - 2110 E-Mail: a.wehrm...@centersystems.com -- Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
