Re: [Sofia-sip-devel] Registration authentication problem - 904 No matching challenge
Hi Laurent, 2011/8/1 Laurent Lecigne lleci...@gmail.com: I am currently stuck trying to get a Sofia SIP agent to properly register to a registrar (Open IMS core). Whenever the agent gets challenged it does not reattempt to provide its credentials : the registration keeps failing. Basically, a nua_authenticate() is triggered on nua_r_register 401 but Sofia says there is no matching challenge (904) although: o realm (enclosed within double quotes), login and password provided are accurate, o same operation handle is used than received event (hope this is the way to go). I expected the framework to resubmit REGISTER after this nua_authenticate() call (i.e. I am not re-registering after). Sofia-SIP does not support AKA authentication out-of-the-box. What you need is a custom authentication client plugin. I should probably write an example how to do it... -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Sofia-Sip Supported RTP Stack
Hi, 2011/8/18 Meftah Tayeb tayeb.mef...@gmail.com: do sofia include a RTP stack ? how can i integrate existing RTP stack with Sofia ? Please see sofsip-cli for an example of integrating a media stack with Sofia SIP. and if RTP stack is integrated/supported, what codecs are supported ? can i inject aditional codecs to the existing RTP Stack ? All the codecs that do not require any fancy SDP features are readily supported. -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] register time-out
Hi, 2011/8/19 Nick Knight n...@omniis.com: I have a client which registers against a server fine. The server imposes a register at 3600 which turns up in the expires field. Sofia then tries to register at aruond 1800, which is too much before what the server will allow. How is this controlled? Why is sofia ignoring it? The 3600 seconds is the maximum lifetime of the expiration. Sofia SIP follows the practice outlined in most IETF documents to do the refresh roughly at half the expiration time. If you want Sofia SIp to register once in hour, propose expiration time of 7200 seconds? -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Invite with Multipart content
Hi Harinath, There is no decent API for sending MIME multipart from nua at the moment. Currently, if nua notices that there is a payload, it won't touch it. I have some doubts regarding encoding of multipart (iow, it won't work.) I have some ideas about the API, like including an empty SDP within multipart or so, but unfortunately little time to work on them. Patches are always welcome, if you feel adventurous. --Pekka 2011/6/15 harinath nanpaly harin...@globalwirelesstech.com: I need to send INVITE message with multipart content (application/sdp + application/femtointerfacemsg). I would appreciate if you could give some ideas on it. I am having hard time making the sdp body manually, which eventually goes into the MIME payload. Any ideas? Thanks, Harinath -- EditLive Enterprise is the world's most technically advanced content authoring tool. Experience the power of Track Changes, Inline Image Editing and ensure content is compliant with Accessibility Checking. http://p.sf.net/sfu/ephox-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ 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 remove Supported header from outgoing request.
Hi Inca, 2011/6/22 Inca Rose incar...@gmail.com: I'm using NUA and I see that by default the Supported header is sent with timer, 100rel I want to control the supported header contents and find out that NUATAG_SUPPORTED only adds supported values, and only SIPTAG_SUPPORTED can override it. I want to be able to not send Supported headers at all if the UA doesn't support any extension. I set SIPTAG_SUPPORTED_STR() but the stack sends the Supported header empty. There is any way to tell the stack not to send the Supported header if it is empty ? You should be able to get rid of Supported with SIPTAG_SUPPORTED_STR((void *) -1)). -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] 'Network error' on FreeBSD: how to debug?
Hi Yuri, 2011/6/19 Yuri y...@rawbw.com: On 06/18/2011 17:50, Yuri wrote: recvfrom call retrieving result for DNS query never receives the result for some reason. I found bug in code resolving the sip-dig hang: in file sres_blocking.c lines if (c-block-fds[i].revents | POLLERR) sres_resolver_error(c-resolver, c-block-fds[i].fd); if (c-block-fds[i].revents | POLLIN) sres_resolver_receive(c-resolver, c-block-fds[i].fd); should look like this: if (c-block-fds[i].revents POLLERR) sres_resolver_error(c-resolver, c-block-fds[i].fd); if (c-block-fds[i].revents POLLIN) sres_resolver_receive(c-resolver, c-block-fds[i].fd); Thanks for catching this... After this sip-dig works fine. But empathy still has 'Network error'. Does your UA send DNS queries? Do they look valid? Where Sofia reads the resolv.conf? Can you get logs from UA? -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Segmentation Fault After BYE 407 Proxy Authentication
Hi Jerry, 2011/6/23 Jerry Richards jerry.richa...@teotech.com: One more thing on this. The segmentation fault was due to the assert() function crashing in my phone. This was not caused by sofia-sip. But the additional comparison to sip_method_bye should be added to sofia-sip so the assert() does not cause a printf. I'm afraid that does not work (in other words, the ACK sent by nua is gibberish if the cr is BYE, because it has CSeq number different from INVITE.) Thanks for reporting this, I'll try to make a fix to the root cause. -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] ipv4/ipv6 issues
Hi Florian, 2011/6/29 Florian Limberger florian.limber...@aon.at: ... When an ipv6 client initiates a call to a ipv4 client the call is properly established!! If ipv4 calls ipv6, the final ACK is not sent. ... Probably, this has something to do with the Contact header, which contains the transport address which has been used for registration. However, if ipv6 invites ipv4, everything is fine. Note, that the ipv4 only host has a link-local ipv6 address but has no ipv6 connectivity to the Internet. Please advise me where and how the transport for an ACK is determined and where in the code this can be fixed. You are correct when you assume that the Contact header is in play. It determines the transport for ACK, unless there happens to be some Record-Route headers. Your proxy is supposed to add those Record-Route headers in case it notices that caller is IPv4 and callee IPv6. Alternatively, the IPv4-only User Agent could try to do something clever, like still using the same proxy to send ACK it used to send INVITE, if it cannot use the transport from Contact header. Unfortunately, Sofia is not so clever. -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] auto nua_ack with bad UIR (without destination port)
Hi Ivan, 2011/7/13 Иван Чистяков zetru...@gmail.com: I send INVITE to SLFphone and recv responce 100/180/200 with bad Contact header (without port). And sofia stack trying to send auto ACK with this bad Contact header (without port). SLFphone can use non standart port, but stack trying to use default port 5060. How to fix it? ACK sip:softphone@172.16.0.129:3296 SIP/2.0 You can try to force Sofia SIP to use a certain address for next hop, for instance, include NUTAG_PROXY(sip:172.16.0.129:3296) in nua_invite() or nua_ack() or nua_respond() tags. -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] PRACK and 200 OK(INVITE) forking issue
Hi Nauman, 2011/7/1 Nauman Sulaiman nauman762-h...@yahoo.co.uk: We are handling all media (SDP) ourselves and have set PRACK to be one of the methods our app handles. We are using the nua API. Can Pekka or someone say whether PRACK and forking is supported with the way we are using the Stack. We've seen some posts saying there are various bugs. When PRACK is not used the first 200OK received is passed to the application with cancel sent to the others. This is the behaviour we would like even when PRACK is used. However it seems that when the first 183 reliable is received this results in subsequent 200OK from other forks to be discarded. We just want the first 200OK final response to be passed to our application irrespective of which fork its from and are happy for CANCEL to be sent to others. Unfortunately PRACK (and early dialog) support is broken vis-a-vis the forking. Sofia ditches the original invite, and treats the first early dialog as the only one. -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] bug fix in su_time0.c for MacOS X 64bit
2011/7/11 Frode Isaksen frode.isak...@bewan.com: There is a bug in su_time0.c for MacOS X 64bit since the field tv_sec in struct timeval is 64bits instead of 32bits as in su_time_t, so you cannot cast su_time_t to struct timeval. This patch fixes this: Thanks. I've applied the patch. -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Problem with sofia codec selection
Hi, 2011/7/12 Olivier Deme od...@druidsoftware.com: Is it possible to configure sofia to pick a selection of codecs based on the SDP offered by the network and the SDP specified in nua_respond, rather than picking a single codec? Use SOATAG_RTP_SELECT(SOA_RTP_SELECT_COMMON). -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] tport_check_trunc() issue
Hi Arsen, 2011/7/13 Arsen Chaloyan achalo...@yahoo.com: Although this is a typical race condition issue, it still seems pretty harmless as the goal of this function is to set tp-tp_trunc which isn't used currently. To get rid of this message, I have just commented out tport_check_trunc() but wanted to know where tp-tp_trunc was supposed to be used. As far as I recall, it was supposed to be used to streamline the receiving code path. Now we ask kernel for datagram length and then try to receive it, I believe using the MSG_TRUNC flag was supposed to remove one system call from that. -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Possible Memory Leak if INVITE was not succesfully.
Hi Stefan, 2011/7/13 Stefan Opfermann sip.rie...@yahoo.de: we have an application based on Sofia SIP 1.12.11 which uses directly INVITE methods between two SIP User Agents without a REGISTER server. In the case that the User is still busy or the caller not allowed to call ( 404 Not found etc. ), we have noticed a memory leak on the caller side if the outgoing INVITE event (nua_r_invite) has an status unequal 200 OK. We tried to generate the ACK Answer by our Event Handler, so NUTAG_AUTOACK( 0 ) and also the automatic answer with NUTAG_AUTOACK( 1 ) option. The ACK is generated correctly , but in booth modi we lost memory. In case of an error response (status = 300) the ACK response is part of the transaction and transaction layer takes care of sending it. The ACK Response Event nua_r_ack is hidden them from application. It is just used to send nua_ack() request to stack. Application is not supposed to see it. So the question is, where is the best place to free the handle ? When you get final error response in nua_r_invite or nua_i_state with nua_callstate_terminated? -- Pekka.Pessi mail at nokia.com -- Get a FREE DOWNLOAD! and learn more about uberSVN rich system, user administration capabilities and model configuration. Take the hassle out of deploying and managing Subversion and the tools developers use with it. http://p.sf.net/sfu/wandisco-d2d-2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] tls_connect() Invoked On Every Outbound INVITE
Hi Jerry, 2011/6/9 Jerry Richards jerry.richa...@teotech.com: Anyone know why sofia-sip would re-establish a TLS connection for every outbound INVITE? That's what I see happening. Logs are shown below, after calling nua_invite() while a TLS connection was already establised (Note: you'll probably see some extra logs I added to help debug this). The tport_by_addrinfo() requires that the TLS connection was opened with same canon name (in this case, [FD00::2A0:25FF:FE00:2ABD]) or that the subjects from the certificate match your canon name (looks like your subject is 200.21.3.10). Was the already open connection inbound? --Pekka Thanks, Jerry --[452] nua: nua_invite: entering --[453] nua(0xbb04a8): sent signal r_invite --[458] nua(0xbb04a8): recv signal r_set_params --[459] nua: nua_stack_set_params: entering --[460] nua(0xbb04a8): event r_set_params 200 OK --[461] nua: nua_application_event: entering --[465] nua(0xbb04a8): recv signal r_set_params --[466] nua: nua_stack_set_params: entering --[467] nua(0xbb04a8): event r_set_params 200 OK --[468] nua: nua_application_event: entering --[472] nua(0xbb04a8): recv signal r_invite --[473] nua: nua_stack_set_params: entering --[474] nua(0xbb04a8): adding session usage --[482] nta_leg_tcreate(0xbb2fd0) --[483] [2]outgoing_create() --[484] outgoing_create: [1]route_url-us_url-url_host=[FD00::2A0:25FF:FE00:2ABD] --[485] outgoing_create: [2]route_url-us_url-url_host=[FD00::2A0:25FF:FE00:2ABD] --[486] outgoing_create: [3]route_url-us_url-url_host=[FD00::2A0:25FF:FE00:2ABD] --[487] outgoing_create: [4]route_url-us_url-url_host=[FD00::2A0:25FF:FE00:2ABD] --[488] outgoing_create: [5]route_url-us_url-url_host=[FD00::2A0:25FF:FE00:2ABD] --[489] nta: selecting scheme sip --[490] url-url_host=[FD00::2A0:25FF:FE00:2ABD] --[491] us-us_url-url_host=[FD00::2A0:25FF:FE00:2ABD] --[492] tport_tsend(0xba7390) tpn = */[FD00::2A0:25FF:FE00:2ABD]:5061 --[493] tport_resolve addrinfo = [fd00::2a0:25ff:fe00:2abd]:5061 --[494] tport_by_addrinfo(0xba7390): not found by name */[FD00::2A0:25FF:FE00:2ABD]:5061 --[495] tport_tls_connect: Entering... --[496] tport_alloc_secondary(0xba7390): new secondary tport 0xbd29d0 --[497] tls_init_secondary: SSL_new(ctx=0xbaf2d8) --[498] ...SSL_new() returned ssl=0xbbe820 --[499] tls_init_secondary: SSL_set_bio(ssl=0xbbe820, rbio=0xbb0198, wbio=0xbb0198) --[500] tls_init_secondary: SSL_set_connect_state(ssl=0xbbe820) --[501] tls_init_secondary: SSL_set_mode(ssl=0xbbe820, SSL_MODE_ACCEPT_MOVING_WRITE_BUFFER) --[502] [IPv4]tport_base_connect(TCP): bind(s=29, len=16, sin_fam=2, sin_port=50451=0xc513, sin_addr=0x0) --[503] [IPV6]tport_base_connect(TCP): bind(s=29, len=16, sa_fam=2, sin6_port=50451=0xc513, --[504] sin6_flowinfo=0, sin6_scope_id=2, sa_data=00 00 00 00 00 00 00 00 28 25 bd 00 43 25 bd 00 --[505] tport_tls_connect(0xbd29d0): bind(local-ip): Invalid argument --[506] [IPv4]tport_base_connect(TCP): connect(s=29, len=28, sin_fam=10, sin_port=50451=0xc513, sin_addr=0x0) --[507] [IPV6]tport_base_connect(TCP): connect(s=29, len=28, sa_fam=10, sin6_port=50451=0xc513, --[508] sin6_flowinfo=0, sin6_scope_id=2, sa_data=fd 00 00 00 00 00 00 00 02 a0 25 ff fe 00 2a bd --[509] tport_tls_connect(0xbd29d0): connecting to tls/[fd00::2a0:25ff:fe00:2abd]:5061/sips --[510] tport(0xbd29d0): reset timer --[511] tport_queue(0xbd29d0): queueing 0xbaff08 for tls/[fd00::2a0:25ff:fe00:2abd]:5061 --[512] nta: sent INVITE (13472340) to */[FD00::2A0:25FF:FE00:2ABD]:5061 --[513] tport_pend(0xbd29d0): pending 0xbaff08 for tls/[fd00::2a0:25ff:fe00:2abd]:5061 (already 0) --[514] nta: timer set to 32000 ms --[515] nua(0xbb04a8): call state changed: init - calling, sent offer --[516] nua(0xbb04a8): event i_state INVITE sent --[517] nua: nua_application_event: entering --[530] tls_connect(0xbd29d0): events CONNECTING --[531] tls_connect(0xbd29d0): events NEGOTIATING --[532] tls_connect(0xbd29d0): events NEGOTIATING --[533] SSL_get_peer_certificate(ssl=0xbbe820) --[534] ...SSL_get_peer_certificate() returned cert=0xbbf3b8 --[535] SSL_get_verify_result(ssl=0xbbe820) returned 0 (success) --[536] NeedCert: TRUE --[537] NeedCRL: FALSE --[538] NeedOCSP: FALSE --[539] OCSP_URL: --[540] tls_post_connection_check(0xbd29d0): Peer Certificate Subject 0: 200.21.3.10 --[541] goto X509_VERIFY_OK: NeedOCSP=FALSE --[542] tport_send_event(0xbd29d0) - ready to send to (tls/[fd00::2a0:25ff:fe00:2abd]:5061) -- -- EditLive Enterprise is the world's most technically advanced content authoring tool. Experience the power of Track Changes, Inline Image Editing and ensure content is compliant with Accessibility Checking. http://p.sf.net/sfu/ephox-dev2dev ___
Re: [Sofia-sip-devel] NUTAG_WITH and BYE
Hi Vladimir, 2011/6/9 Vladimir Luchko vlad.l...@mail.ru: Can I answer to the incoming BYE with 200 ok from the application without using NUTAG_WITH? I`ve init nua with NUTAG_AUTOANSWER(0) and NUTAG_APPL_METHOD(BYE). Than I got nua_i_bye event. I want to answer 200 with nua_resopnd(nh, 200, NULL, NUTAG_NULL()), but stack emits nua_i_error 500 - answer to non-existing request. The idea is that there can be multiple transactions ongoing (e.g., INVITE and PRACK) at the same time. why I must use NUTAG_WITH to respond to BYE? What side effects can I get if doing something like that: if (!t (sr-sr_method == sip_method_invite || sr-sr_method == sip_method_bye)) break; If your application receives INVITE, then BYE, and decides to answer to BYE before answering to INVITE, stack might get confused (answering INVITE instead and leaving BYE server transaction hanging around). -- Pekka.Pessi mail at nokia.com -- EditLive Enterprise is the world's most technically advanced content authoring tool. Experience the power of Track Changes, Inline Image Editing and ensure content is compliant with Accessibility Checking. http://p.sf.net/sfu/ephox-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] one nua, multiple accounts
Hi Luca, 2011/6/1 Luca Olivetti l...@ventoso.org: BTW, the documentation says that sofia-sip will generate the contact using the provided NUTAG_M_USERNAME, an equal sign and a random string, but it doesn't seem to be the case, so I didn't bother checking for the presence of an equal sign and stripping the rest in sip-sip_request-rq_url-url_user. The documentation tries to be future-proof in case nua will grow automatic multi-registration support. Ok, in that case, url_user will be the one supplied in NUTAG_M_USERNAME or will it be the complete one? In the latter case, is there a function to strip the auto-generated part or should I roll my own? It will contain = and then some random data. You can strip it away with strchr(user, '=') etc. -- Pekka.Pessi mail at nokia.com -- Simplify data backup and recovery for your virtual environment with vRanger. Installation's a snap, and flexible recovery options mean your data is safe, secure and there when you need it. Discover what all the cheering's about. Get your free trial download today. http://p.sf.net/sfu/quest-dev2dev2 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] one nua, multiple accounts
Hi Luca, 2011/5/29 Luca Olivetti l...@ventoso.org: If in nua_register I use NUTAG_M_USERNAME, can I be sure that the proxy will use it as a Contact: for every incoming request? If that's the case, I no longer need to rely on the To: header. Yes, NUTAG_M_USERNAME is best way to implement an user agent with multiple registrations. The proxy is supposed to use the URL from contact. Of course, there may be buggy proxies. -- Pekka.Pessi mail at nokia.com -- Simplify data backup and recovery for your virtual environment with vRanger. Installation's a snap, and flexible recovery options mean your data is safe, secure and there when you need it. Data protection magic? Nope - It's vRanger. Get your free trial download today. http://p.sf.net/sfu/quest-sfdev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] No ACK sent after PRACK and CANCEL
Hi Inca, 2011/5/20 Inca Rose incar...@gmail.com: We are experiencing a strange behavior in the following scenario: A: Using Sofia SIP stack A --- INVITE A -- 183 with RSeq A PRACK A --- OK ( PARCK ) A --- 180 A -- CANCEL A -- OK ( Cancel ) A -- 487 Request terminated At this point I was expected A to send ACK to terminate the INVITE Transaction, but the ACK is never sent. In the same scenario, but without PRACK, the ACK after the 487 This is a known bug There seems to be serious problems in processing the final answer from another fork in case of early dialogs. Thanks for reporting this. -- Pekka.Pessi mail at nokia.com -- vRanger cuts backup time in half-while increasing security. With the market-leading solution for virtual backup and recovery, you get blazing-fast, flexible, and affordable data protection. Download your free trial now. http://p.sf.net/sfu/quest-d2dcopy1 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] No ACK sent after PRACK and CANCEL
Hi, 2011/5/24 Pekka Pessi ppe...@gmail.com: 2011/5/20 Inca Rose incar...@gmail.com: We are experiencing a strange behavior in the following scenario: A: Using Sofia SIP stack A --- INVITE A -- 183 with RSeq A PRACK A --- OK ( PARCK ) A --- 180 A -- CANCEL A -- OK ( Cancel ) A -- 487 Request terminated At this point I was expected A to send ACK to terminate the INVITE Transaction, but the ACK is never sent. In the same scenario, but without PRACK, the ACK after the 487 This is a known bug There seems to be serious problems in processing the final answer from another fork in case of early dialogs. Thanks for reporting this. Hmmm, do you happen to use NTATAG_UA(0)? -- Pekka.Pessi mail at nokia.com -- vRanger cuts backup time in half-while increasing security. With the market-leading solution for virtual backup and recovery, you get blazing-fast, flexible, and affordable data protection. Download your free trial now. http://p.sf.net/sfu/quest-d2dcopy1 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] MSRP Support
Hi, 2011/5/3 Ubuntu Explorer ubuntuexplo...@gmail.com: Are there any plans to add MSRP support to Sofia SIP in the roadmap? We might add support for the SDP negotiation for setting up MSRP, but I see no MSRP implementation ahead. However, feel free to contribute. ;) -- Pekka.Pessi mail at nokia.com -- What Every C/C++ and Fortran developer Should Know! Read this article and learn how Intel has extended the reach of its next-generation tools to help Windows* and Linux* C/C++ and Fortran developers boost performance applications - including clusters. http://p.sf.net/sfu/intel-dev2devmay ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Early dialog not working after Update method
Hi Matteo, 2011/5/17 Matteo mbrancale...@voismart.it: going on with debug, the issue is that outgoing_find does not check for different to_tag in provisional response. I've created a patch that works for me which handles different to_tag in provisional responses and manage RSeq in right way if to_tag changed. but I don't know if this is the proper way to do it. The early dialog and especially early media with forking is, indeed, tricky. There was some talk with Anthony M. on media handling in that case, but we got never around to define how the media should behave. (Should the early media from all forks be played? Should they be mixed? If a 200 OK is received from another fork, how it should be handled?) Currently, I think, nua on FreeSwitch sticks to first fork sending it a 100rel response as you noticed. My original idea was to handle each fork (especially with reliable 1xx responses) in a separate nua handle and let the application (in this case FreeSwitch) to handle all the hairiness related to media, but perhaps there is a cleaner approach. -- Pekka.Pessi mail at nokia.com -- What Every C/C++ and Fortran developer Should Know! Read this article and learn how Intel has extended the reach of its next-generation tools to help Windows* and Linux* C/C++ and Fortran developers boost performance applications - including clusters. http://p.sf.net/sfu/intel-dev2devmay ___ 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 know remote hold in nua_i_invite event？
Hi Chris, 2011/4/24 Chris hlidea...@gmail.com: In sofia-sip nua reference page, it indicates we can retrieve tags SOATAG_ACTIVE_AUDIO() when we get a nua_i_invite event. It is useful if I could use that tag to check remote holding events. However, I found I can not get SOATAG_ACTIVE_AUDIO() in nua_i_invite event. I can only get its value in nua_i_state. My question is: Is there any possibility to check remote holding in nua_i_invite? In my design, I want to handle SDP negotiation in nua_i_state but all SIP related things in their respective events. The current Sofia SIP version does not include the media-related tags in nua_i_invite, however, that seems to be a bug. I've pushed a patch fixing that into gitorious: https://gitorious.org/sofia-sip/sofia-sip/commit/4f58617329fa8821ba1014a27e32c9292cf6f874/diffs -- Pekka.Pessi mail at nokia.com -- WhatsUp Gold - Download Free Network Management Software The most intuitive, comprehensive, and cost-effective network management toolset available today. Delivers lowest initial acquisition cost and overall TCO of any competing solution. http://p.sf.net/sfu/whatsupgold-sd ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] NUTAG_DETECT_NETWORK_UPDATES still unimplemented under linux?
Hi Luca, 2011/4/18 Luca Olivetti l...@ventoso.org: A2 = MD5(REGISTER:sip:ekiga.net) auth_response: db79b8a5046c76654b742311b7d4a33d = MD5(53060310bbad90036d54e1954983ae4e:4dab21dbe3306f1513e34e9e24eedc00a7827bd9394f:14bd7ed1967527d3770ab1f7c12901c5) (qop=NONE) svd: nta.c: 7785: outgoing_create: Assertion `tport_name_is_resolved(orq-orq_tpn)' failed. It seems the same as this one: http://www.mail-archive.com/sofia-sip-devel@lists.sourceforge.net/msg03243.html outgoing_create is using a bogus override_tport, here, around line 7770 of nua.c, causing the crash The network change is probably suffering from bit-rot. The transport is cached in nua/outbound.c, I suppose it should drop its reference to tport_t, too, when network change is detected. See: https://gitorious.org/~ppessi/sofia-sip/pessi-sofia-sip/commits/network-change -- Pekka.Pessi mail at nokia.com -- Benefiting from Server Virtualization: Beyond Initial Workload Consolidation -- Increasing the use of server virtualization is a top priority.Virtualization can reduce costs, simplify management, and improve application availability and disaster protection. Learn more about boosting the value of server virtualization. http://p.sf.net/sfu/vmware-sfdev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Handling 486 Busy Here
Hi, 2011/4/10 Manuel Argüelles manuel.arguel...@gmail.com: I'm using an nta agent in a multi-thread application (proxy), I noticed that during an invite transaction if the peer returns a 486 Busy Here code, sofia sip seems to internally handle it and replies with an ACK, since the callback function doesn't gets called, is there a way to get a notification about it? The ACK to error responses such as 486 is part of the transaction so NTA takes care of it. What kind of notification would you like to get? -- Pekka.Pessi mail at nokia.com -- Xperia(TM) PLAY It's a major breakthrough. An authentic gaming smartphone on the nation's most reliable network. And it wants your games. http://p.sf.net/sfu/verizon-sfdev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Setting sofia-sip Source Port
Hi Jerry, 2011/4/4 Jerry Richards jerry.richa...@teotech.com: This is a Govt. requirement (AS-SIP). We can't change it. Huh? And I thought that IMS had it bad... What function is invoked when reconnect is attempted if the initial connection fails? Hm. socket(), bind() and connect()? --Pekka -Original Message- From: Pekka Pessi [mailto:ppe...@gmail.com] Sent: Monday, April 04, 2011 9:14 AM To: Jerry Richards Cc: sofia-sip-devel@lists.sourceforge.net Subject: Re: [Sofia-sip-devel] Setting sofia-sip Source Port Hi Jerry, 2011/4/4 Jerry Richards jerry.richa...@teotech.com: I temporarily got it working by patching the tport_tls_connect() function to set susa.su_port to 5061 (instead of setting it to 0). I do use nua, so I'll try NUTAG_OUTBOUND() as you suggest. We have a requirement that TLS connections must specifically use source port 5061 (not a random port). There is the problem you cannot reconnect once the initial connection fails (as there is no way to make difference between old and new connections). Please try to reconsider the requirement, it will not work. -- Pekka.Pessi mail at nokia.com -- Pekka.Pessi mail at nokia.com -- Xperia(TM) PLAY It's a major breakthrough. An authentic gaming smartphone on the nation's most reliable network. And it wants your games. http://p.sf.net/sfu/verizon-sfdev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Integrating sofia with other event loop libraries
Hi, 2011/3/30 Leandro Lucarella llucare...@integratech.com.ar: Hi, I noticed sofia has its own event loop and there is integration with glib event loop. It is possible (and easy) to integrate sofia with other event loop libraries? I'm interested in integrating it with libev specifically. Integration is possible, the easiness depends how well the concepts suit with the other event loop libraries. See various su_*_port.c files for examples. If not (possible or easy ;), is it sofia thread-safe? Sofia-SIP is threadsafe in the sense it uses no static data. Can I enqueue sofia events into my own loop and thus do the processing in another thread (calling sofia API from another thread, different from where sofia event loop is running)? The NUA API is supposed to do suit that model, iow, you should be able to call nua methods from multiple threads. Other APIs have some design problems. Your idea of queuing the events sounds interesting. ;) -- Pekka.Pessi mail at nokia.com -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Setting sofia-sip Source Port
Hi, 2011/3/31 Jerry Richards jerry.richa...@teotech.com: I am setting my contact address to specify port 5061 (for TLS), but a wireshark trace shows the message being sent with a source port of 1025. This is confusing the Redcom server. How can I make sofia-sip send the message with the same port as is in the contact header? Do you use nua? If so, you can set NUTAG_OUTBOUND(outbound natify use-rport), and Sofia will use the ephemeral source port in Contact, too. Otherwise, the situation is a bit more complicated. What kind of setup you have? -- Pekka.Pessi mail at nokia.com -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Setting sofia-sip Source Port
Hi Jerry, 2011/4/4 Jerry Richards jerry.richa...@teotech.com: I temporarily got it working by patching the tport_tls_connect() function to set susa.su_port to 5061 (instead of setting it to 0). I do use nua, so I'll try NUTAG_OUTBOUND() as you suggest. We have a requirement that TLS connections must specifically use source port 5061 (not a random port). There is the problem you cannot reconnect once the initial connection fails (as there is no way to make difference between old and new connections). Please try to reconsider the requirement, it will not work. -- Pekka.Pessi mail at nokia.com -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] NOTIFY sip frag set to 503 when receiving 407 Invite response
Hi, 2011/3/30 Nauman Sulaiman nauman762-h...@yahoo.co.uk: Hi, when performing an attended transfer, when the transfer destination sends a 407 to the INVITE with replaces generated by transferee, a NOTIFY with 503 sipfrag is sent to the transferrer. Code is here in nua_session.c in function nh_referral_respond. The transfer destination is in different domain. The resulting NOTIFY 503 sent to the transferrer makes him think the transfer has failed rather than just being in progress. Is there some reason for this or is it a bug? I think it is a bug, the stack should check if the request is restartable. --Pekka -- Pekka.Pessi mail at nokia.com -- Create and publish websites with WebMatrix Use the most popular FREE web apps or write code yourself; WebMatrix provides all the features you need to develop and publish your website. http://p.sf.net/sfu/ms-webmatrix-sf ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] release debug version of test_nua.exe run differently (sofia-sip-1.12.11)
Hi Allen, 2011/3/25 Allen alleng...@gmail.com: Thanks a lot for your reply. I have tried the --no-nat setting, the results are the same. Actually the log information I provided in my original email for Visual studio 2003 release version is already the result with BOTH --no-nat AND --no-proxy setting, otherwise it halts even earlier with the following log information: OK. The problem is, test_nua test cases may suffer from some nasty timing issues, but the newer check-based tests do not work at all on Windows. TEST NUA-2.6.1: REGISTER b to c test_nua: testing b_call-nh = nua_handle(b-nua, b_call, (tag_type_t)0, (tag_value_t)0) test_nua: ok: (b_call-nh = nua_handle(b-nua, b_call, (tag_type_t)0, (tag_value_t)0)) nta: REGISTER (10166550): Connection refused (10061) with tcp/[192.168.1.5]:1685 I have test these release version of test_nua.exe on XP SP3, Vista business, and Windows 7 Home Preminum, all all have same results as the log shown in my original email. All these three operation systems have the latest patches installed. I have checked the su_init() function, it's the su_init(void) at the line 304 of su.c, it DOES include the code to load the _DisconnectEx function pointer, and I have checked the log information, the loading function call didn't fail. OK, so that is not an issue. I am confused why the release version runs differently from debug version, the debug version just passes all the test cases. The optimization has a tendency to change the work flow slightly and exposing weaknesses and bugs not present in unoptimized versions. Of course, the compilers always have bugs, most of them show up when optimizing. I wonder if the latest MSVCs allow you to debug optimized versions. If so, please try to increase optimization level in debug version and try to see if that helps. According to my limited experiences, it might caused by not initializing some varibles before using them since the debug version always set the varibles to zero when they are declared. Compiler should warn about not initializing some variables. The project files disable quite a many warnings on MSVC, please see if enabling them reveals something. Would you please give me some more advices to dig into this issue? Thanks! First, try test_nua with -s option. If it works with that, the issue is most probably in the test suite. You can also run test_nua with various logging options, using -e with --log=a, --log=b or --log=c might reveal something. I'm mostly interested in the logs from the runs that stop at test case 9.1.4 ... -- Pekka.Pessi mail at nokia.com -- Enable your software for Intel(R) Active Management Technology to meet the growing manageability and security demands of your customers. Businesses are taking advantage of Intel(R) vPro (TM) technology - will your software be a part of the solution? Download the Intel(R) Manageability Checker today! http://p.sf.net/sfu/intel-dev2devmar ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] release debug version of test_nua.exe run differently (sofia-sip-1.12.11)
Hi Allen, 2011/3/23 Allen Guan alleng...@gmail.com: I am new to this mail list. I have built sofia-sip-1.12.11 under XP SP3 and tried the test programs. I have noticed strange behaviors of the test program test_nua.exe when running in release version. I have tried to compiled with Visual studio 2003, visual studio 2005 and visual studio 2010 express. All the debug versions of test_nua.exe compiled with these three compilers run well; but the release versions hang in the middle of the running, I have attached the log information got when I ran these programs in verbose mode. Please take a look at it and give me some advices on the possible reasons why these release versions have issues and how to tune them up to make them work. The test_nua has some problems in the way some WinSock implementations route the local TCP/IP traffic. You can try to run it with command line option --no-nat, and see if the problem persists. You could also check if the su_init() manages to load the _DisconnectEx function pointer. I was under the impression that it should work in later XP SPs, too, but it is not first time I have made a mistake. -- Pekka.Pessi mail at nokia.com -- Enable your software for Intel(R) Active Management Technology to meet the growing manageability and security demands of your customers. Businesses are taking advantage of Intel(R) vPro (TM) technology - will your software be a part of the solution? Download the Intel(R) Manageability Checker today! http://p.sf.net/sfu/intel-dev2devmar ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Version 1.12.11
Hi Jerry, 2011/3/23 Jerry Richards jerry.richa...@teotech.com: I noticed the latest trunk still has the following line in the configure.ac file: AC_INIT([sofia-sip], [1.12.10devel]) Shouldn't this line show 1.12.11? It seems to me that I have actively forgotten about CVS. I committed the release to CVS, too. -- Pekka.Pessi mail at nokia.com -- Enable your software for Intel(R) Active Management Technology to meet the growing manageability and security demands of your customers. Businesses are taking advantage of Intel(R) vPro (TM) technology - will your software be a part of the solution? Download the Intel(R) Manageability Checker today! http://p.sf.net/sfu/intel-dev2devmar ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Getting started: parsing a message
Hi Bendan, 2011/3/21 Brendan Loudermilk bren...@apwit.com: As a weekend project I've begun to play around with Sofia-SIP. I'm using FFI to build a simple Ruby interface to some of the lower-level functions in the library. My first goal is to be able to parse arbitrary strings from a Ruby networking library. I saw the example on this page for parsing memory, but I'm wondering if theres a simpler way. What kind of strings you want to parse? Something like Contact: sip:192.168.1.2:5060? As a secondary question, what options are there for those of us who are interested in the SIP/SDP protocol behavior implementations, but want to implement our own networking? Are we forced to re-implement the various behavioral RFCs in our implementation? Or is there an event-based hook portion of Sofia that I have yet to discover? I'm a bit loss what you want to do by yourself? Interface towards the sockets and network? Something else? -- Pekka.Pessi mail at nokia.com -- Enable your software for Intel(R) Active Management Technology to meet the growing manageability and security demands of your customers. Businesses are taking advantage of Intel(R) vPro (TM) technology - will your software be a part of the solution? Download the Intel(R) Manageability Checker today! http://p.sf.net/sfu/intel-dev2devmar ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Prack
Hi Vikas, 2011/3/18 Vikas Bhat vikasbhat0...@gmail.com: i am implenting early dialog in Sofia sip application using prack. the application is stateless using NTA modul. my application send invite with Supported header,receives 183 with RSeq. then the applcation tries to send the PRACK message as shown below but it fails orq=nta_outgoing_prack(leg, //leg created before sending the invite (nta_outgoing_t *)dialogue-orq, // this is orq of INVITE message send earlier using nta_outgoing_tcreate (nta_response_f *)sofia_sip_response, //call back function for receiving response (nta_outgoing_magic_t*)dialogue, //dialogue pointer that is created after sending initial INVITE NULL, NULL, NTATAG_STATELESS(1), SIPTAG_RACK(pRackHdr), //added rack header TAG_END()); Kindly guide me to implement Prack . If you want to use nta_outgoing_prack(), you first have to create a new instance of INVITE client transaction specific to the new early dialog you want to PRACK with nta_outgoing_tagged(). When you receive an early response with Require: 100rel, you create the new invite transaction with nta_outgoing_tagged(), and keep both it and the original invite transaction around. Depending on your application, you might run to a situation where the final 200 OK is received from some other UA (or you might receive 100rel responses from other UAs, too). There is a disturbing @bug in nta_outgoing_tagged(), however, I believe the leak is fixed, but the PRACK and early dialog code is not very well tested. Please report if you run to any assert() failure. -- Pekka.Pessi mail at nokia.com -- Enable your software for Intel(R) Active Management Technology to meet the growing manageability and security demands of your customers. Businesses are taking advantage of Intel(R) vPro (TM) technology - will your software be a part of the solution? Download the Intel(R) Manageability Checker today! http://p.sf.net/sfu/intel-dev2devmar ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Good TLS Hellos But No SIP REGISTER Sent
Hi Jerry, 2011/3/22 Jerry Richards jerry.richa...@teotech.com: I am seeing an issue with TLS where the REGISTER message is not getting sent. The TLS Hello/Certificate exchange seems to always happens correctly (which just precedes the REGISTER request). The scenario is that it is previously registered and then later lose registration because of this issue. Interestingly enough, it seems to be timing related, because if I enable all the sofia debug logging in the sofia stack, it does not have this problem. Could this be a timing issue? Any thoughts on how to resolve this? How you manage your TLS connections? Do you have an active connection during the lifetime of the registration? Can you send other requests, like, INVITE after a period of inactivity? -- Pekka.Pessi mail at nokia.com -- Enable your software for Intel(R) Active Management Technology to meet the growing manageability and security demands of your customers. Businesses are taking advantage of Intel(R) vPro (TM) technology - will your software be a part of the solution? Download the Intel(R) Manageability Checker today! http://p.sf.net/sfu/intel-dev2devmar ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] soatag_local_sdp_str_ref undeclared ('soa:remove duplicate definitions')
Hi Erik, 2011/3/24 Erik Habicht e.habi...@thiesen.com: Looks like there was a bogus soatag_session_sdp_str_ref declaration instead of soatag_local_sdp_str_ref. Yes ... adding SOFIAPUBVAR tag_typedef_t soatag_local_sdp_str_ref; also solve the problem. Is SOFIAPUBVAR tag_typedef_t soatag_session_sdp_str_ref; needed any more? In my case removing SOFIAPUBVAR tag_typedef_t soatag_session_sdp_str_ref; works. I think I've planned on adding SOATAG_SESSION_SDP() at some point and the duplicate was an abortive attempt on it. Anyways, thanks for spotting this, I've already pushed the patch to gitorious. -- Pekka.Pessi mail at nokia.com -- Enable your software for Intel(R) Active Management Technology to meet the growing manageability and security demands of your customers. Businesses are taking advantage of Intel(R) vPro (TM) technology - will your software be a part of the solution? Download the Intel(R) Manageability Checker today! http://p.sf.net/sfu/intel-dev2devmar ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] Anyone still using CVS?
Hello all, Are there a a lot of people still using CVS? As it is now, CVS is not really very useful in tracking the development, nor it is very useful in tracking releases as Jerry noticed. If there is no objections, I'll stop updating the CVS repo and try to remove the links to it from the Sourceforge developer pages. -- Pekka.Pessi mail at nokia.com -- Enable your software for Intel(R) Active Management Technology to meet the growing manageability and security demands of your customers. Businesses are taking advantage of Intel(R) vPro (TM) technology - will your software be a part of the solution? Download the Intel(R) Manageability Checker today! http://p.sf.net/sfu/intel-dev2devmar ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] ANN: release candidate 1 for sofia-sip-1.12.11
Hello all, Long time no release. However, now I've put out a pre-release tar package available from sofia-sip.org: http://sofia-sip.org/~ppessi/sofia-sip-1.12.10pre11rc1.tar.gz md5sum: 3c9dbb7dee430ccb5d1d0b088eef5516 sha1sum: a0da65f9e8031df1f016545dc83b9ff362760b35 sha256sum: 6c683625cf9c069c6dbe9c99f29377c123555200deeaab359a37a2684d917ff6 The release candidate is tagged in http://gitorious.org/sofia-sip/sofia-sip as sofia-sip-1_12_11rc1 with git commit 93ccbc1230708aa4947d824823e39ce56105291f I've only tested it on Ubuntu 10.10 and Windows 7 (compiled with VS 2010), I'd appreciate if people with more exotic platforms would give it a try. -- Pekka.Pessi mail at nokia.com -- What You Don't Know About Data Connectivity CAN Hurt You This paper provides an overview of data connectivity, details its effect on application quality, and explores various alternative solutions. http://p.sf.net/sfu/progress-d2d ___ 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 retrieve the local bound SIP port when created dynamically? (as used in contact-header)
Hi, 2011/1/31 EiSl 1972 eisl1...@gmail.com: I have the feeling the answer lays in front of my nose, but for some reason I cannot find/see it... nua_get_params(.., NTATAG_CONTACT(NULL), TAG:END()) and look for ntatag_contact in the tag list of nua_r_get_params(). I'm using the released stable version 1.12.10. I know there is a lot of incident repair and other development still ongoing. If I would like to use a more recent version, which branch/tag is considered then most stable? The master branch at git://gitorious.org/sofia-sip/sofia-sip.git is currently pretty stable... -- Pekka.Pessi mail at nokia.com -- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! Finally, a world-class log management solution at an even better price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires February 28th, so secure your free ArcSight Logger TODAY! http://p.sf.net/sfu/arcsight-sfd2d ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Sofia Getting 900 Internal Error When SIPTAG_TO_STR(sip:7348902; rfrid=eaff000000221c6035b5@200.21.3.10)
Hi Jerry, 2011/2/3 Jerry Richards jerry.richa...@teotech.com: Okay, I think I fixed it. For some reason, sofia did not like when I set both NUTAG_URL() in the nua_handle() call and also set SIPTAG_TO_STR() to the same thing in the nua_invite() call. So I removed specifying the SIPTAG_TO_STR() in the nua_invite() call and the error no longer occurs. I think your problem is a syntax violation: SIPTAG_TO_STR(sip:7348902;rfrid=eaff00221c6035b5@200.21.3.10) Try to enclose the URI in , otherwise To header parser thinks the URI is sip:7348902 and the parameter rfrid=eaff00221c6035b5@200.21.3.10 has invalid syntax.. -- Pekka.Pessi mail at nokia.com -- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! Finally, a world-class log management solution at an even better price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires February 28th, so secure your free ArcSight Logger TODAY! http://p.sf.net/sfu/arcsight-sfd2d ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] [sofia-sip-devel] digest authorization using HA1 directly
Hi Paulo, 2011/1/13 Paulo Vicentini vicentini.pa...@gmail.com: Attached is hack to use HA1 with sofia-sip Example: nua_authenticate(nh, SIPTAG_EXPIRES_STR(3600), NUTAG_AUTH(authentication), TAG_END()) where authentication is string: Digest-HA1:realm:user:HA1 I did it in a more complex way, so you should have HA1+Digest:realm:user:HA1+hash For example HA1+Digest:ims3.so.noklab.net:user1:HA1+c0890ff7a4fadc50c45f392ec4312965 -- Pekka.Pessi mail at nokia.com -- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! Finally, a world-class log management solution at an even better price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires February 28th, so secure your free ArcSight Logger TODAY! http://p.sf.net/sfu/arcsight-sfd2d ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Sip messages truncated
Hi Philippe, 2011/1/21 Philippe Maymat pmay...@keyyo.com: Since I allowed th multidialog sent in Notify, I received fragmented messages and it seems that the message is truncated by th sofia SIP stack because ethereal show all informations. Check the Content-Length from the ethereal, is it correct? It would help if you could save the problematic message from ethereal and send it on the list, too. -- Pekka.Pessi mail at nokia.com -- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! Finally, a world-class log management solution at an even better price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires February 28th, so secure your free ArcSight Logger TODAY! http://p.sf.net/sfu/arcsight-sfd2d ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Multi home and sofia-sip
Hi Inca, 2011/1/21 Inca Rose incar...@gmail.com: What about the selection of Contact and Via and Transport on a multi-home environment ? I remember that the selection was not consistent, for example sending the packet from NIC1 while selecting Via / Contact from NIC2. This is a problem that has no generic solution, at least not in Linux. Can you bind sockets or applications to a NIC in iOS? Also there is no way to check from which NIC the proxy is reachable, imagine a VPN situation where the proxy is only reachable from the VPN but not from the real NIC. Where can I add iOS implementation for the Network change detection ?? See nua_register.c/nua_network_changed_cb() and su_os_nw.c and su_root_add_network_changed() -- Pekka.Pessi mail at nokia.com -- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! Finally, a world-class log management solution at an even better price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires February 28th, so secure your free ArcSight Logger TODAY! http://p.sf.net/sfu/arcsight-sfd2d ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Format of contact header
Hi Stefan, 2011/1/13 Stefan Eckel stefan.ec...@c4b.de: we are looking for a method to put transport tags into the contact header field of an invite message. With nua_invite(m_handle, SIPTAG_CONTACT_STR( sip:100.100.100.100:5160;tansport=tcp ) , we get Contact: sip: 100.100.100.100:5160;transport=tcp If there is no aroung the URI, it ends at first ;. Try SIPTAG_CONTACT_STR( sip:100.100.100.100:5160;tansport=tcp ) -- Pekka.Pessi mail at nokia.com -- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! Finally, a world-class log management solution at an even better price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires February 28th, so secure your free ArcSight Logger TODAY! http://p.sf.net/sfu/arcsight-sfd2d ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Multi home and sofia-sip
Hi Inca, 2011/1/14 Inca Rose incar...@gmail.com: Hi; I remember there was a limitation with Sofia-SIP and multi-home hosts ( in my case is a device ). I do not remember if it was the Via header not populated correctly or the transport was not selected correctly. There was any improvement in this area ?? Why am I asking ? I need to support network changing on the fly during an active call. If the device changes network, from 3g to wifi or vice-versa, during an active call, the application has to be notified and do whatever needed to continue the call without user intervention. If NUA was initialized with the 3G address how can I move the call to WiFi -- I need to change the Contact and SDP, so I need to send a re-invite/update, but NUA is not aware of the WiFi transport. There was some code in NUA and NTA for changing the transports (grep for NUTAG_DETECT_NETWORK_UPDATES). However, it is pretty untested and it is currently triggered only in OS X. Triggering it on Linux should be trivial, but it has been on TODO list last five years.. ;-/ -- Pekka.Pessi mail at nokia.com -- Special Offer-- Download ArcSight Logger for FREE (a $49 USD value)! Finally, a world-class log management solution at an even better price-free! Download using promo code Free_Logger_4_Dev2Dev. Offer expires February 28th, so secure your free ArcSight Logger TODAY! http://p.sf.net/sfu/arcsight-sfd2d ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Link error on visual studio 2008
Hi, 2010/12/28 Philippe Maymat pmay...@keyyo.com: I found a solution, the file string.c is not included in the project and contains all missing definitions. Oops, the dll projects were not updated. I've added them to the git version. Now my problem is a crash on line 550 in nua_client.c when subscribing. The same code works fine on linux, so I try to find the difference ... Any news? --Pekka -- Pekka.Pessi mail at nokia.com -- Gaining the trust of online customers is vital for the success of any company that requires sensitive data to be transmitted over the Web. Learn how to best implement a security strategy that keeps consumers' information secure and instills the confidence they need to proceed with transactions. http://p.sf.net/sfu/oracle-sfdevnl ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Memory allocation problem on windows, Sorry It s nua_stack.c file :)
Hi Philippe, 2010/12/29 Philippe Maymat pmay...@keyyo.com: I made several modifications to compile cotrrectly the SOFIA-SIP project with Visual studio 8. There is mistakes on both project files for static and dynamic library on line 4305 and 3216. There was /files/files instead of /files. With this modification the XML is valid and the project cn be loaded. Thanks for reporting this, I've fixed the problem. I need to add smoothsort.c in static lib project which is forgotten too. ...and added this one, too. Now it works but It seem I have a big memory leak, software crash during subscribe... My project works well on linux (QT project) and it is the same revision of SOFIA-SIP I use th version from this repository : git://gitorious.org/sofia-sip/sofia-sip.git Is it correct ? Because I do not had problem with the old cvs version but this doesn't work for me on windows ... Now I try to debug and find why it crashes, but it's very hard ... I often have error in nta.c line 2388 because of a corrupted content in strucures. I have this problem in both static and dynamic lib linking Can you run the test cases? The check.cmd script should run them. -- Pekka.Pessi mail at nokia.com -- Gaining the trust of online customers is vital for the success of any company that requires sensitive data to be transmitted over the Web. Learn how to best implement a security strategy that keeps consumers' information secure and instills the confidence they need to proceed with transactions. http://p.sf.net/sfu/oracle-sfdevnl ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] [sofia-sip-devel] digest authorization using HA1 directly
Hi Paulo, 2010/12/18 Paulo Vicentini vicentini.pa...@gmail.com: Thanks for your tips If It uses a Digest-HA1 scheme, it would need to receive the same from the registrar server (matching schemes). It would be good to be server agnostic (Digest) int ca_credentials(... ... ... if ((scheme != NULL !su_casematch(scheme, ca-ca_scheme)) || (realm != NULL !su_strmatch(realm, ca-ca_realm))) return 0; It needs to somehow to ignore above code, maybe checking for Digest-HA1 Yes, the ca_credentials should somehow magically match the Digest-HA1 with Digest, and somehow store the HA1 in the auth_client_t structure so that the digest algorithm itself would recognize the HA1. The current authentication client design does not support that very well. -- Pekka.Pessi mail at nokia.com -- Gaining the trust of online customers is vital for the success of any company that requires sensitive data to be transmitted over the Web. Learn how to best implement a security strategy that keeps consumers' information secure and instills the confidence they need to proceed with transactions. http://p.sf.net/sfu/oracle-sfdevnl ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] [sofia-sip-devel] digest authorization using HA1 directly
Hi Paulo, 2010/12/15 Paulo Vicentini vicentini.pa...@gmail.com: I using sofia-SIP as an UAC to register with a SIP registrar I'd like to avoid using the secret (it is indeed not available) directly while creating a digest authorization header with: int auc_digest_authorization(auth_client_t *ca, su_home_t *home, char const *method, url_t const *url, msg_payload_t const *body, msg_header_t **return_headers) Only HA1 = md5(username:realm:password) is available So that I intend to use HA1 = md5(username:realm:password) instead What do you say about that? It is doable with some modifications to iptsec/auth_client.c. You could modify the ca_credentials to store only the HA1 in the ca_client_t structure instead of the password (in case of Digest) and add a special scheme, e.g., Digest-HA1 where the password would contain the HA1. Patches are welcome. -- Pekka.Pessi mail at nokia.com -- Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Populating to header from Refer message with replaces
Hi Nauman, 2010/12/8 Nauman Sulaiman nauman762-h...@yahoo.co.uk: Hi, using sofia 1.12.10 how to populate TO header of Invite created in response to incoming REFER with refer-to header containing replaces parameter. I can't just use the refer-to uri on its own as it has the all the replaces stuff. I suppose i could grab the various bits from the refer-to header and create a To uri but was wondering if there is a better way. I am currently using the following tags SIPTAG_FROM_STR SIPTAG_TO_STR NUTAG_URL NUTAG_NOTIFY_REFER NUTAG_REFER_EVENT the new invite is being created with a replaces header, just the to header is wrong Is there something extra in To header? I guess you could parse the refer-to URL and remove all the header parameters from it. Do you want to have the to-tag from Replaces in the new INVITE? The Replaces header and to-tag is used to find an existing call to drop when the new referred call gets connected. -- Pekka.Pessi mail at nokia.com -- Lotusphere 2011 Register now for Lotusphere 2011 and learn how to connect the dots, take your collaborative environment to the next level, and enter the era of Social Business. http://p.sf.net/sfu/lotusphere-d2d ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] we *still* have a race condition
2010/10/23 Jen Chitty jenchi...@vtech.ca: We are trying to track down an intermittent failure in the Sofia SIP stack where we appear to be getting an EPERM error back from some POSIX API call or another on a powerpc Linux 2.6.23. This error is somehow triggering the stack to report an ACK timeout, even though no timer has expired. Does this sound familiar to anyone? Anyway, I came across a line of code in tport_type_udp.c that I found somewhat alarming: su_soerror(self-tp_socket); /* XXX - we *still* have a race condition */ Does anyone know what race condition this is referring to? When network reports an error with ICMP message, the kernel stores the error in the socket and wakes up the application with special error event. Application can ask for the error code using su_soerror() (getsockopt system call with SO_ERROR argument) but if it does not, next time application tries to read or write from the socket, the error is reported to it. If Sofia SIP application talks with two other SIP instances, say A and B, and it sends a request to A, but A is unreachable and A's router reports back an error with ICMP, send to *B* may fail if the ICMP packet from A is received before call to su_vsend() (sendmsg() system call). The above code clears the error just before call to su_vsend. Doing so reduces the size of the window where an unrelated ICMP can trigger failure when sending but it does not remove it entirely. -- Pekka.Pessi mail at nokia.com -- Centralized Desktop Delivery: Dell and VMware Reference Architecture Simplifying enterprise desktop deployment and management using Dell EqualLogic storage and VMware View: A highly scalable, end-to-end client virtualization framework. Read more! http://p.sf.net/sfu/dell-eql-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] make check fails
Hi Filippo, 2010/9/20 Della Betta Filippo filippo.dellabe...@telecomitalia.it: Attached config.h and config.status. Below output of ifconfig... It seems to me that you do not have openssl-dev, but my make check passes through even if I remove it... Mysterious. -- Pekka.Pessi mail at nokia.com -- Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing. http://p.sf.net/sfu/novell-sfdev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Using VPN
Hi all, 2010/9/16 mikhail.zabal...@nokia.com: Hi, using 1.12.10. Sofia doesn't seem to select the vpn ip when connected over VPN. It still seems to be listening on the usual WiFI access point ip, this is the same no matter what ip is passed to NUTAG_URL. It should work if the nua stack is created with this tag. I set it as an url_t equivalent to sip:IPADDR:*. The socket is bind with maddr parameter, like sip:IPADDR:*;maddr=IPADDR. That said, it is unfortunate that Sofia cannot determine the preferred IP binding by itself, even though it requires platform-specific code (so a Linux fix using rtnetlink will not solve the problem for iPhone). Yes, the netlink or equivalent code is surely needed. -- Pekka.Pessi mail at nokia.com -- Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing. http://p.sf.net/sfu/novell-sfdev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] su_root_threading query
2010/9/15 Nauman Sulaiman nauman762-h...@yahoo.co.uk: Hi, using 1.12.10 i've set su_root_threading to multithreading mode. But when i call nua_register for example from the main thread of my app it blocks it, it appears to be all the sresolv stuff. Should this be expected and i should be calling the nua functions on a separate thread. If someone could confirm. The nua_register should just queue a message for the other thread to process. Even in single-threaded mode, it should queue it and not block (but the message would be processed only when the tread returns to su_root_run() or does su_root_step()). -- Pekka.Pessi mail at nokia.com -- Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing. http://p.sf.net/sfu/novell-sfdev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] make check fails
2010/9/17 Della Betta Filippo filippo.dellabe...@telecomitalia.it: From a clean VM Ubuntu 10.04 x86, I did the following steps to reproduce failure sudo apt-get install build-essential autoconf automake1.9 libtool git-core check git clone http://git.gitorious.org/sofia-sip/sofia-sip.git cd sofia-sip ./autogen.sh mkdir build cd build ../configure make make check I could not reproduce the problem on my host. Could you send me your config.h and config.status (and perhaps output from ip addr)? -- Pekka.Pessi mail at nokia.com -- Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing. http://p.sf.net/sfu/novell-sfdev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] make check fails
Hi Filippo, 2010/9/10 Della Betta Filippo filippo.dellabe...@telecomitalia.it: With the latest git code, make check fails during check_nua (I have 4/5 errors). The first test that fails is call_2_3_1. So I launched check_nua call_2_3_1 With CHECK_NUA_THREADING=no : fails With CHECK_NUA_THREADING=yes : ok Can you check this ? It runs fine on my Ubuntu Lucid host. What kind of environment you have? -- Pekka.Pessi mail at nokia.com -- Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing http://p.sf.net/sfu/novell-sfdev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Minor patches
Hi Filippo, 2010/8/27 Della Betta Filippo filippo.dellabe...@telecomitalia.it I’m trying to update sofia-sip to the latest git version. Is it possible to have the function nua_handle_by_call_id exported ? (nua_h.patch) I've applied your patches but slightly modified latter two. Thanks. --Pekka -- Pekka.Pessi mail at nokia.com -- This SF.net Dev2Dev email is sponsored by: Show off your parallel programming skills. Enter the Intel(R) Threading Challenge 2010. http://p.sf.net/sfu/intel-thread-sfd___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Problem in using sofia-sip libraries
Hi Amir, 2010/8/14 Amir Khezrian amir.khezr...@gmail.com: From the errors, It seems that compiler can't find sofia-sip/msg_types.h and sofia-sip/msg_mime.h Have i missed something that is necessary to be installed before using sofia-sip? You should -I the directory where the inlude files has been installed, not the sofia-sip source tree. By default the include path is /usr/local/include/sofia-sip . -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] forking calls and overlapping
2010/8/12 Михаил Кривушин krivushi...@gmail.com: Hello! We use FreeSWITCH and OpenSIPS, and have problem - when we try use forking calls, it seems that sofia nta/nta.c doesnt distinct INVITE-s by Via headers, and its branches. Is it right, or I need more code reading? By default, the server side on nta.c detects the branched INVITEs (and responds with 482 Request Merged to them). On client side, the same client side transaction can be used to receive all the responses. It is also possible to create a fork-specific client-side transaction with nta_outgoing_tagged(). The nua call model does not support forking, and the sdp processing and media handling gets outright confused on multiple forks. -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Internal error at nua_stack.c:2388 on SUBSCRIBE
Hi Philippe, 2010/8/13 Philippe Maymat pmay...@keyyo.com with the latest git repo, I now have the following error in return of nua_subscribe: Internal error at nua_client.c:550 It's due to the ';', with any other char, it works. Could you show us your code? How do you initialize nh? Which other tags you include with nua_handle() and nua_subscribe() besides NUTAG_URL? -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] sofia-sip not sending correct Via headers in 200 OK to NOTIFY request
Hi Gaurac, 2010/8/13 Pekka Pessi ppe...@gmail.com: 2010/8/11 Gaurav Srivastva gaurav...@yahoo.com: The Via headers on Linux are sent correctly if TPORT_LOG is not defined. If I define this variable then the problem starts happening. Oh my. Thanks for digging this out. I've pushed the fix to gitorious, please test. -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ 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_callstate_authenticating don't work
Hi Paolo, 2010/8/5 Paulo Pizarro paulo.piza...@gmail.com: The nua_callstate_authenticating is documented: http://sofia-sip.sourceforge.net/refdocs/nua/nua__tag_8h.html#904045132b398207f1597320c860eca3 But, it is not used on the source (only defined). Maybe, it's better to remove it from the enum or fix documentation with TODO. It seems to me that it never has been used. I'll fix the documentation. -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] One more question about cloned task
Hi Pete, Hi Pete, 2010/8/7 Pete Kay pete...@gmail.com: Do you have multiple nta_agent_t instances running, or how do you handle the UAS side? What kind of task you plan to implement in the cloned task, some kind of db lookup? Yes, I am using DB to store routing information. Have you considered using su_msg_send() and su_msg_reply() to talk with the DB thread instead of multi-threaded agent? Please note that while nta functions are thread-safe, the nta objects are not. If you want to see later messages from UAC, you need to insert a Record-Route header into the request which you forward towards the UAS. I am looking at http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__record__route.html which shows how to creat a record_route structure. How do I add it to the outgoing sip message? Which api should I use? That depends on how you plan to do the forwarding. E.g., sip_record_route_t *rr = sip_record_route_format(msg_home(msg), ...); msg_header_insert(mg, NULL, (msg_header_t *)rr); Also, should I store both UAC-related RR and UAS-related RR in the RR-URI's user part? User part is easiest, I think. In the hash table you suggested, should I use an UUID to represent each leg ( one for UAC and the other one for UAS) and then store the leg corrspond to each UUID? I would not use legs: if you process same request twice (say, one of your users forwards his calls to another user) the legs cannot make difference between loops. I'd process each incoming request either statelessly or with default leg, and look for the UUID from the topmost route. Also, I don't understand why it matter whether I am using one or two nta_agent_t. I still need two UUID ( one for each leg ) regardless of whether I use one or two nta_agent_t right? Or am I missing something? Well, in principle you could do with one UUID and RR if you used one agent. You have to remove the resulting Route headers when processing the later messages, too. Which API can I use to remove the Route header and when you say later messages, what are you referring to? When should the Route header be removed? The RFC 3216 sections 16.4 and 16.12 describes how the Route header should be processed. Basically, when you receive a request with Route header, you should forward request based on topmost Route instead of request-URI. If topmost Route URI belongs to your proxy, you remove it. The lr URI parameter complicates matters, however. -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Null Callid and To Tag
2010/8/5 Vikas Bhat vikasbhat0...@gmail.com: Thanks for the information. Do u mean to say that Sofia Stack(NTA) should send the 400. since we are using the STACK in is STATELESS mode .should our application check for CallID with null values and respond with 400. Yes and no, the nta stack should check for bad requests even in stateless mode. is there any stack configuration to be enabled from SOFIA STACK side to Check such scenerios and respond appropriately. See, e.g., the documentation of NTATAG_BAD_REQ_MASK(): http://sofia-sip.sourceforge.net/refdocs/nta/nta__tag_8h.html#23d0ebce6f3a594c5547e0ff72fc5777 -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] One more question about cloned task
Hi Pete, 2010/7/30 Pete Kay pete...@gmail.com: In my request_handler, I am creating a new cloned_task upon receiving of INVITE from uac. I am using this cloned_task to create a leg to talk to the uas and forward the INVITE to the uas. Do you have multiple nta_agent_t instances running, or how do you handle the UAS side? What kind of task you plan to implement in the cloned task, some kind of db lookup? When I get another message from uac, such as ACK, how can I reuse the same cloned task and the same leg that was previously created or I should recreate it for ACK as well? If you want to see later messages from UAC, you need to insert a Record-Route header into the request which you forward towards the UAS. Where can I store the cloned task and the nta_leg_t object and have it be reused for the next message received from the same UAC? The problem with legs is that you have no control on Call-ID and From/To tags, they are chosen by UAC/AUS. Also, a dialog may loop through your proxy multiple times. It is better to insert a Record-Route URL that you can use to lookup the dialog and the associated state from some kind of hash table. (E.g., insert a hashed identifier of your dialog in user-part of the RR URI, use that to lookup from the hash table). If the RR points to different nta_agent_t instances, you need to insert two Record-Route headers, one used by UAC, another by UAS. With two Record-Route headers you can also easily make difference between requests sent by original UAC and UAS. You have to remove the resulting Route headers when processing the later messages, too. -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Multiple transports with single nua_create instance
Hi, 2010/8/2 Nauman Sulaiman nauman762-h...@yahoo.co.uk: Hi, using sofi 1.12.10 we have managed t support multiple registrations using separate nua handle to nua_register call, single nua_create call. However is it possible to specify separate ports and transport for each nua_handle OR do we need to create a separate stack instance to support this ie multiple nu_create calls with unique NUTAG_URL for each instance. Unfortunately, you can only specify two transports in nua_create(), one with NUTAG_URL() another with NUTAG_URL_SIPS(). -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Fixed documentation of the detailed client call model
2010/8/5 Paulo Pizarro paulo.piza...@gmail.com: I'm sending a patch fixing the documentation of the detailed client call model. Thanks, applied. I hope this is true! :) Hope so too. ;) -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Using TLS with sofia
2010/8/2 Nauman Sulaiman nauman762-h...@yahoo.co.uk: Hi, using version 1.12.10. We've compiled OpenSSL 0.9.8, also compiled the 2 source files related to tls in tport. We know that in NUTAG_URL we must pass in transport=tls. What else do we need to do to enable tls on Sofia, is there any sample code to show how to use it? You should have root (cafile.pem) and server certificates ready (agent.pem) in a suitable directory ($HOME/.sip/auth by default). -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Make an app they can't live without Enter the BlackBerry Developer Challenge http://p.sf.net/sfu/RIM-dev2dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] adding rinstance to NOTIFY request line in nua
2010/7/22 Markus Bucher buch...@in.tum.de: Is the notify for a existing subscribe/notify session or is it for a blind notify? It's actually for both, though blind notifies have priority. The scenario is, that there is a security server between my application an the receiver of the notify (client). This server adds the rinstance to the contact-header of a register. I save this rinstance-tag and have to add it to the request line of all requests to the client (that are initiated by my application). If a packet is not in a dialog and does not have the rinstance tag, the packet will be droped by the server. I'm nor really able to try it, so I don't really know if subscribe/notify sessions are concerned, but blind notifies are definitely concerned. NUTAG_URL() should work for blind NOTIFYs but not with the SUBSCRIBEd NOTIFYs. The latter take the URL from the Contact (or Record-Route) header. However, they are part of an existing dialog, so your security servers should let them through. (You may need to respond to SUBSCRIBE before sending NOTIFY, however.) -- Pekka.Pessi mail at nokia.com -- The Palm PDK Hot Apps Program offers developers who use the Plug-In Development Kit to bring their C/C++ apps to Palm for a share of $1 Million in cash or HP Products. Visit us here for more details: http://p.sf.net/sfu/dev2dev-palm ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] NTA authentication request problem
2010/7/24 Mayur Mahajan mayurmahajan...@gmail.com: I have made a shorter version of my mail. When I send a request to the server through my NTA client, I get the response of the message(407 Unauthorized) in the callback. So, when the callback function gets the 407 message it invokes authorize_ua() method. The problem is I am not able to send the request which authorizes me. Could you please help me in the following things :- The 407 Unauthorized is sent by a proxy (as opposed to the 401 Unauthorized which is sent by UAS). The 407 carries Proxy-Authenticate header (instead of WWW-Authenticate in 401) and the subsequent retry should contain Proxy-Authorization header. nta_leg_destroy(context-c_leg), context-c_leg = NULL; if (context-c_leg) This is an another problem... -- Pekka.Pessi mail at nokia.com -- The Palm PDK Hot Apps Program offers developers who use the Plug-In Development Kit to bring their C/C++ apps to Palm for a share of $1 Million in cash or HP Products. Visit us here for more details: http://p.sf.net/sfu/dev2dev-palm ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Null Callid and To Tag
2010/7/26 Vikas Bhat vikasbhat0...@gmail.com: Sofia stack is not able to detect empty/null CallID in INVITE request. similarly Sofia is not able to detect empty/NULL TO TAG. Ideally Sofia should reject the calls in above scenerios. I think NTA is supposed to return a 400 response to a request received from network if CallID header is missing. On client side, it generates a CallID automatically, if application has not provided one. Likewise, it generates a random To tag if application has not provided one. The To tag is always empty or missing in initial requests; do you propose that a Sofia should automatically try to clear any call if a response to an INVITE request contains empty/null To tag? -- Pekka.Pessi mail at nokia.com -- The Palm PDK Hot Apps Program offers developers who use the Plug-In Development Kit to bring their C/C++ apps to Palm for a share of $1 Million in cash or HP Products. Visit us here for more details: http://p.sf.net/sfu/dev2dev-palm ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Threading - Root problem Newbie
Hi Rohan, 2010/7/24 rohan kala rohankal...@gmail.com: 1) I have initialized su_root_create() function in my main() function, now do I have to initialize it in callback function or every othe function to send request or incoming request etc.. Or I have to just put su_root_run(context-c_root) every time when I expect callback function. The main loop is executed within the su_root_run() function (until one of your callbacks call su_root_break(context-c_root) 2) What's the role of su_root_threading() where should I initialize it? If you create a su_root_t clone (e.g., nua_create() creates one), it can be executed with the original thread and su_root_t main loop, or by an separate thread. You can select the mode with su_root_threading() call before the clone is created. 3) Once I have created nta_outgoing_tcreate() function and had sent request with it do I have to do nta_outgoing_destroy() before using nta_outgoing_tcreate() etc. Or threading can take care of it? There can be multiple client transactions ongoing, you should call nta_outgoing_destroy() after you get the final response (or, in case of INVITE, after you send the ACK to 2XX response). -- Pekka.Pessi mail at nokia.com -- The Palm PDK Hot Apps Program offers developers who use the Plug-In Development Kit to bring their C/C++ apps to Palm for a share of $1 Million in cash or HP Products. Visit us here for more details: http://p.sf.net/sfu/dev2dev-palm ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Contact header incorrect
Hi, 2010/7/26 Nauman Sulaiman nauman762-h...@yahoo.co.uk: Hi, using Sofia 1.12.10. I have multiple sip accounts, with separate nua handles for each nua_register etc. When i get a nua_i_invite and send a response it seems to pick a contact randomly from sip accounts that i have for the contact header. I am switching on the sip_request field to determine the account the invite is for then i manually attach SIPTAG_CONTACT_STR to each nua_respond etc then all is ok. But is there a better way i should be doing this? Unfortunately, no. SIPTAG_CONTACT() is best you can do at the moment. -- Pekka.Pessi mail at nokia.com -- The Palm PDK Hot Apps Program offers developers who use the Plug-In Development Kit to bring their C/C++ apps to Palm for a share of $1 Million in cash or HP Products. Visit us here for more details: http://p.sf.net/sfu/dev2dev-palm ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Does soa module support “Del ayed Offer” ?
2010/7/21 edson.gomes.leme edson.gomes.l...@uol.com.br: Hi Pekka Pessi; a) Does the Sofia-SIP soa module support “Delayed Offer” ? Yes.. b) How can this call scenario be implemented with the soa module? SOFIA-SIP (soa) A (UAC) B (UAS) | | |-- INVITE --| | | |--- 100 Trying -| |--- 180 Ringing | | | |--- 200 (offer) | |-- ACK (answer) | | | You need nothing special (but the usual remote sdp tags in nua_i_invite are missing). c) How can this call scenario be implemented with the soa module? SOFIA-SIP (soa) A (UAC) B (UAS) | | |-- INVITE --| | | |--- 100 Trying -| |--- 180 Ringing | | | |--- 200 (offer) | |-- ACK (answer) | | | Currently, this requires a kludge, you have to include an empty payload (I think SIPTAG_PAYLOAD_STR() is ok) in nua_invite(). --Pekka -- Pekka.Pessi mail at nokia.com -- The Palm PDK Hot Apps Program offers developers who use the Plug-In Development Kit to bring their C/C++ apps to Palm for a share of $1 Million in cash or HP Products. Visit us here for more details: http://p.sf.net/sfu/dev2dev-palm ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Multiple Registration support
Hi, 2010/7/12 Nauman Sulaiman nauman762-h...@yahoo.co.uk Then i tried with 2 different accounts, one Sipgate the other Callcentric. I created 2 separate register handles and separate call back handlers for each handle. I call nua_register twice with the separate handles now. It should work fine, there is probably something else hosed in your code.. What i noticed is the REGISTER messages go out to the 2 different providers and i get a 401 back from each one, but my callback handlers are not getting called, its like the stack is not passing these on. I also noticed that the REGISTER messages to the 2 differnet providers are getting sent with the same CSEQ number!!! So i assume the stack is getting completely confused with the responses. I assumed that sending nua_register with separate handles would allow each REGISTER transaction to be treated separately and the responses would be routed to the separate handlers. The CSeq number is generated based on the current time. Two REGISTERs may have same CSeq as long as their Call-ID header (and tags in From and To headers) are different. Can someone give me a high level description how to set up what i want to, as mentioned i have a single account working fine. It should work just fine, but you can try to enable more detailed debugging logging with NUA_DEBUG=9 and NTA_DEBUG=9 environment variable. -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Sprint What will you do first with EVO, the first 4G phone? Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] [PATCH] Fix gai_strerror re-declaration in MinGW
2010/7/8 Stefano Sabatini ssabat...@reilabs.com: Exactly, this way the patch is much simpler. Thanks! Applied. -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Sprint What will you do first with EVO, the first 4G phone? Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] [PATCH] Fix gai_strerror re-declaration in MinGW
2010/7/6 Stefano Sabatini ssabat...@reilabs.com: Hi, as in subject. The patch requires to run autoheader to re-generate the config.h.in file. Thanks for patch. I wonder if it is enough just to #undef gai_strerror? No need to check its declaration? -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Sprint What will you do first with EVO, the first 4G phone? Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Still no answer to DNS problems
2010/7/2 Inca Rose incar...@gmail.com: I dont know if this only happen to me or I'm not using a configuration flag that is document, but I have a very hard time figuring this out. ... The problem is that the next REGISTER goes again to 1.2.3.4. There is no mechanism that nta can remember that 1.2.3.4 is not answering and put it at the bottom of the list and continue to use 1.2.3.5, the last one that did respond ? In other words, how to force the stack to use from now 1.2.3.5 and not 1.2.3.4 ??? We would need blacklisting/graylisting for the 1.2.3.4. The current code implements graylisting for SRV records (by manipulating their priority, if I recall correctly). However, there is no priority associated with A records, so we would need a different solution. Perhaps a list (or binary tree or hash table or ...) of bad IP addresses/ports/protocols? -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Sprint What will you do first with EVO, the first 4G phone? Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first ___ 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: initializing SIP stack failed
2010/6/30 Stefano Sabatini ssabat...@reilabs.com: Hi all, I'm running this simple code in Windows Vista: nta: master transport created tport(00F27958) to */*:*/sip tport(00F27958): calling tport_listen for udp tport(00F27958): new primary tport 00F28128 tport(00F27958): bind(pf=2 udp/[192.168.0.1]): No such file or directory nta: bind(*:*;transport=*): No such file or directory nua: initializing SIP stack failed Can you suggest what the problem may be? For some reason binding a socket to IP address 192.168.0.1 fails. Where the address comes from? Does you Vista box try to share internet connection? If you can compile and run localinfo command, see what it prints? You can also modify NUTAG_URL(sip:stef...@*:*) so it explicitly mentions the IP address, e.g., NUTAG_URL(sip:stef...@*:*;maddr=10.2.3.4) -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Sprint What will you do first with EVO, the first 4G phone? Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] [PATCH] Fix configuration under MinGW
2010/7/6 Stefano Sabatini ssabat...@reilabs.com:
this patch fixes the configuration problem as reported here:
http://thread.gmane.org/gmane.comp.telephony.sofia-sip.devel/3853
Thanks for the patch.
-CFLAGS=$CFLAGS -I\$(top_srcdir)/win32/pthread -DWINVER=0x0501 \
+CFLAGS=$CFLAGS -I${srcdir}/win32/pthread -DWINVER=0x0501 \
Can you actually compile sofia-sip with this? It seems to me that the
relative srcdir does dot work when make descends deeper into source
tree.
--Pekka
--
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Re: [Sofia-sip-devel] I need to send a re-invite with the SDP address set to 0.0.0.0
2010/6/22 Inca Rose incar...@gmail.com: Can someone help me here ? I cannot find a way to set the SDP c= line to 0.0.0.0 I a session update ( re-invite ) I set the SOA_ADDRESS tag to 0.0.0.0 in the reinvite but the new INVITE request keeps sending the c line with the original value. The c=0.0.0.0 is a magic value for call hold, soa does not want to use it. What I'm doing wrong ?? You are not doing anything wrong, but there should be a tag which would allow you to use c=0.0.0.0 to hold a session. -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Sprint What will you do first with EVO, the first 4G phone? Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Is sofia-sip dead?
2010/6/18 Stefano Sabatini ssabat...@reilabs.com: On date Thursday 2010-06-17 01:26:51 -0400, Michael Jerris phoned this: sofia-sip is not dead. The repository is in the process of moving to a new repo. I expect a chunk of patches to be merged in soon, I was discussing this with the maintainer last week. Mike On Jun 16, 2010, at 9:27 PM, Daniel Jabbour wrote: Hi, I am just getting started with a SIP application that I'm writing for a personal project. I am getting ready to use Sofia-SIP as my app's library. However, I noticed: * Sofia-Sip's CVS on SF is 18+ months old, and the latest release in CVS was 1.12.9 * Links on the download page for snapshots, release notes archive, etc are broken * The latest packaged source release is 1.12.10 So, if the latest release is greater than the version tagged in CVS, where is development taking place? Has the repository moved? Is Nokia working entirely in-house? Or is Sofia-SIP dead? Any thoughts would be greatly appreciated. The last darcs commit: Tue Sep 29 14:24:17 CEST 2009 Pekka Pessi first.l...@nokia.com Bug report and original patch by Timo Bruhn. and there hasn't been much activity from the maintainers in the last months and the project somehow didn't managed to get a significant community of developers/contributors. That said, there are many projects depending on sofia-sip, so I hope that there will be enough interest to keep the project alive and possibly make it better, switching to a less obscure SCCS may help. Well, yes, we are currently trying to switch over to git. The git repository is available at http://gitorious.org/sofia-sip The master branch currently contains the more or less the same commits as the darcs repository at sofia-sip.org. It would be useful to mention also that on the sf.net. The next features (and bug fixes) mostly involve DNS resolving and transport handling. There seems to be some nasty bugs in transport handling that might be hard to fix unless some backwards-incompatible changes are made. At the same time it would me nice to get multiprocessing support and perhaps complete the hooks for a SigComp library. -- Pekka.Pessi mail at nokia.com -- This SF.net email is sponsored by Sprint What will you do first with EVO, the first 4G phone? Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Remove an unknown sip header
2010/3/26 Alexandre Brito abr...@av.it.pt: I flowing through sofia sip documentation/code trying to find the best way to remove an unknown sip header. I found that its easy to add any header my using SIPTAG_HEADER_STR or even SIPTAG_HEADER(x) by define a new header struct. However, what's the best option to remove those kind of headers from a sip message? I don't think it is possible when using the tags. Should I create a new SIPTAG around the new header, for instance SIPTAG_P_CHARGING_VECTOR()? What are my options? You have to either go through the list of unknown headers explicitly, or define your own header parser class object. See the sofia-sip-2543 package for examples. If you feel like it, a sofia-sip-3455 package would probably be appropriated by many folks here. -- Pekka.Pessi mail at nokia.com -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] SIP_PAYLOAD_STR first byte 0x00 always causes 0 length content.
2010/3/12 Trevor Nunes trevor.nu...@gmail.com:
The problem is I can send any type of content I want as long as the first
byte is not 0x00 ... If this byte is 0x00 it will send a 0 byte SIP message
out. So for example here is a short message that works perfectly, If I
change the payload[0] = 0x02 line to payload[0] = 0x00 an empty Content
Length is generated ...
--
nua_handle_t *h;
static int lastSequenceNum = -1;
char payload[5];
payload[0] = 0x02;
payload[1] = 0x07;
payload[2] = 0x01;
payload[3] = gMOSMS_seqnum 2;
payload[4] = '\0';
printf(smsagent: Sending L2-ACK with seq: %d \n, gMOSMS_seqnum );
if( lastSequenceNum == gMOSMS_seqnum)
{
printf(smsagent: duplicate seq: %d \n, gMOSMS_seqnum );
}
// send_3ggp2_sms(nua, payload );
h = nua_handle(nua, NULL, SIPTAG_TO_STR(gREMOTE_CLIENT_TAG), TAG_END() );
nua_message(h, SIPTAG_CONTENT_TYPE_STR(application/vnd.3gpp2.sms),
SIPTAG_PAYLOAD_STR( payload ), TAG_END() );
nua_handle_destroy(h);
I guess I need to use a different macro that doesn't treat the payload as a
string or initialize an empty SIP message and cast the pl_data pointer to my
new message ? Should I just malloc() a bunch of bytes and assign this to
pl_data and set pl_en even so I'm still using the SIPTAG_PAYLOAD_STR()
macro so is there way of creating a nua_message where it will treat the
pl_data not as a string ?
Yes, you are correct - you need to use the tag macro SIPTAG_PAYLOAD()
with a pointer to initialized sip_payload_t structure as its argument.
For example,
nua_handle_t *h;
static int lastSequenceNum = -1;
char payload[5];
payload[0] = 0x02;
payload[1] = 0x07;
payload[2] = 0x01;
payload[3] = gMOSMS_seqnum 2;
payload[4] = '\0';
printf(smsagent: Sending L2-ACK with seq: %d \n, gMOSMS_seqnum );
sip_payload_t pl = SIP_PAYLOAD_INIT();
pl.pl_data = payload;
pl.pl_len = sizeof payload;
h = nua_handle(nua, NULL, SIPTAG_TO_STR(gREMOTE_CLIENT_TAG), TAG_END() );
nua_message(h, SIPTAG_CONTENT_TYPE_STR(application/vnd.3gpp2.sms),
SIPTAG_PAYLOAD(pl), TAG_END() );
nua_handle_destroy(h);
--
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Re: [Sofia-sip-devel] simple memory allocation question
2010/3/16 beach...@me.com: Just to make sure I am not doing something wrong, my home gets allocated when my program starts, and I do not destroy it until my (long running) program ends. So although it would not be a true memory leak it sounds like it would cause memory growth if I do not free it, and thus I would be better to free it immediately after nta_incoming_reply(), is that correct ? Yes, that is right. BTW, while we are on the subject of memory allocation, I am also using messages using su_msg_create() to send msgs between my root and clone tasks, where I use the feature to have some user-specified data area allocated with the message, and I never explicitly need to free that memory, correct? Yes, you are correct. The data-area is allocated with a single malloc() along with the main su message. --Pekka -- Pekka.Pessi mail at nokia.com -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] simple memory allocation question
2010/3/15 beach...@me.com: Simple questionwhen I use sip_X_make( home, char *) and then attach that header to a response message I am sending via nta, do I need to explicitly su_free that header? i.e. sip_contact_t *c1 = sip_contact_make( home, szContact1 ) ; nta_incoming_treply( irq, SIP_300_MULTIPLE_CHOICES, SIPTAG_CONTACT( c1 ), TAG_END() ) ; //DO I NEED TO su_free this? Nope, it will be freed when your home is destroyed/unref:ed. If I do need to free it, can I do so immediately after calling nta_incoming_reply? You can do it immediately after calling nta_incoming_reply(). -- Pekka.Pessi mail at nokia.com -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] FYI: Back again
Hello all, As you might have noticed I've not spent much time with Sofia SIP lately. However, Sofia SIP has not been abandoned completely. From now on, I have reserved some time specifically to maintaining Sofia SIP and following this mailing list. -- Pekka.Pessi mail at nokia.com -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] SIP_PAYLOAD_STR first byte 0x00 always causes 0 length content.
2010/3/11 Trevor Nunes trevor.nu...@gmail.com:
I'm trying to send encapsulated 3GPP2 MO-SMS ( CDMA text message ). To do
this the first byte of the payload must be 0x00, if I use SIP_PAYLOAD_STR(
data ) and first byte is zero it seems to truncate assuming the 0x0 is a
NULL terminator.
I've tried using SIP_PAYLOAD( sip_payload_t my_payload ) but not making much
progress. I can successfully generate L2-ACK's since the first byte is
always 0x02.
void send_3gpp2_sms_mo(nua_t *nua, sip_t const *sip)
{
nua_handle_t *h;
sip_payload_t payload = SIP_PAYLOAD_INIT;
char mo_payload[] = {
0x00, 0x00, 0x10, 0x02 // P2P MSG + Teleservice ID
,0x04, 0x02, 0x02, 0x07
,0x62, 0x69, 0x69, 0x69 // Phone number garbled for email!
,0x80, 0x55, 0x01, 0x06
,0x08, 0x38, 0x00, 0x16
,0x20, 0x03, 0x50, 0x0d
,0x03, 0x01, 0x0d, 0x10
,0x0a, 0x50, 0x40, 0x01 // Tele Type
,0x07, 0x0e, 0x09, 0x05 // Callback and other crap
,0x2a, 0x1a, 0x4c, 0x3c
,0x00, 0x80, '\0'
};
fprintf(stderr,smsagent: Tx SMS to phone \n);
h = nua_handle(nua, NULL, SIPTAG_TO_STR(gREMOTE_CLIENT_TAG), TAG_END() );
nua_message(h, SIPTAG_CONTENT_TYPE_STR(application/vnd.3gpp2.sms),
SIPTAG_PAYLOAD( payload ), TAG_END() );
nua_handle_destroy(h);
}
So how can I generate a byte sequence where the first byte is zero and have
it correctly determine the length of the payload? Is there an example of a
direct sip_payload_t .pl_data memcpy or similar that I can get working ?
Probably there are, but all you have to do is to initialize pl_data
with data pointer and pl_len with its length.
--Pekka
--
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Re: [Sofia-sip-devel] suppress Content Disposition
2010/3/8 Ronny Aruch ro...@vocaltec.com: I use sofia 1.12.8. How can I suppress Content Disposition header from being sent. I added SIPTAG_CONTENT_DISPOSITION(NULL)to NUA_INVITE but the header is still sent. It seems to me that there is no way to do that above the API, you have to hack inside the stack. -- Pekka.Pessi mail at nokia.com -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Please, make the sigcomp plugin for sofia-sip free software
2010/3/9 Aleksander Morgado aleksan...@gnu.org: My humble request is then, please make the sigcomp plugin for sofia-sip free software, publicly available for everyone, so that we can keep on developing applications using the sofia-sip stack also including SigComp support. I'll have a peek on the libraries and see if one of them could be used with reasonable effort. However, there might be some legal issues, too, so I can't promise anything. -- Pekka.Pessi mail at nokia.com -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Fw: Changing IP address.
2009/10/2 Robert Han robert...@vtech.ca I just want to raise the issue that we asked you earlier back in July 31 (see below). You said that if we shutdown the sofia stack, and if there are outstanding sip sessions (for example 4 active sessions), we do not need to nua_bye() or even nua_handle_destroy() before invoking nua_shutdown(). according to your reply, the nua stack is guaranteed to terminate outstanding nua operations, and then publish nua_i_terminate as the final event with a operation handle, so that we can destroy that handle. that never happened. That *should* happen. however, the IP address change also means that all the TCP connections and the UDP sockets bound to the old IP address become useless. Do you include maddr= parameter in the NUTAG_URL() in nua_create()? --Pekka -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] 64-bit issue in sofia-sip (patch)
2009/9/24 Aleksander Morgado sofia-sip-de...@aleksander.es: We just faced an issue happening only in 64-bit architecture, and found that it was already fixed by FreeSWITCH guys in their own sofia-sip repo: http://jira.freeswitch.org/browse/SFSIP-136 It seems the fix is not included in the official repo, which would be good to have. Thanks for noticing this. I've applied Stephan's patch. -- Pekka.Pessi mail at nokia.com -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] New release schedule?
2009/9/23 Aleksander Morgado sofia-sip-de...@aleksander.es: Is there any planned date for a new stable (1.12.11) release of sofia-sip? My current plan is to make a next release after we (Maemo devices within Nokia) have stable Maemo 5 release. That should happen Real Soon Now. However, there are few patches by Mikhail Zabaluev, who is responsible for telepathy-sofiasip, that I'd like to include in the stable release. At the same time I'm planning to ditch Darcs as the principal VCS and start using git. I'll have to check if it is possible to host sofia-sip on maemo.org or should I set up something different. -- Pekka.Pessi mail at nokia.com -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Question aboiut NTA and sip_t
2009/9/9 Daniel Corbe dco...@gmail.com: When NTA receives a request which doesn't match an existing transaction it passes a pointer to the callback function of type sip_t. Can this pointer be dereferenced through the lifetime of the transaction or does it go poof as soon as I exit the callback function? As Aleksander said, you can use sip_t as long as the transaction is alive. The sip_t pointer lives inside an abstract msg_t container and you can also get a new reference to it with nta_incoming_getrequest(). After you are done with your sip_t / msg_t reference, you should remove the reference with msg_destroy(). -- Pekka.Pessi mail at nokia.com -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] sresolv: DNS response code 5 (refused) not handled / Fallback to second DNS server not working
2009/9/18 Timo Bruhn voip_v...@web.de:
[...] When sofia-sip tries to register its user agents at startup, the first
dns server is used
to resolve the server name. The server refuses the request. Now the expected
behaviour
should be a fallback to the second server. Unfortunately this never happens.
Yes, the fallback is probably the right thing to do.
One way to fix this is to mark the server as temporarily unavailable as I did
in the patch
attached to this mail. The fix is working fine in my test environment.
Could anyone of the developers please have a look at it to verify that it
does not break
anything else i did not see?
It should not break anything, or at least I don't think it does, but
here is my patch that also falls to the next server on server error
and
not implemented error. Please give it a try, and report if it fixes
your problems (or crashes and burns), too.
--
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--- old-sofia-sip/libsofia-sip-ua/sresolv/sres.c 2009-09-24 13:16:38.0 +0300
+++ new-sofia-sip/libsofia-sip-ua/sresolv/sres.c 2009-09-24 13:16:38.0 +0300
@@ -3503,6 +3503,14 @@
sres_send_dns_query(res, query);
query-q_retry_count++;
}
+ else if (error == SRES_AUTH_ERR ||
+ error == SRES_UNIMPL_ERR ||
+ error == SRES_SERVER_ERR) {
+dns-dns_icmp = res-res_now;
+sres_cache_free_answers(res-res_cache, reply);
+/* Resend query/report error to application */
+sres_resend_dns_query(res, query, 0);
+ }
else if (!error reply) {
/* Remove the query from the pending list */
sres_remove_query(res, query, 1);
--
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Re: [Sofia-sip-devel] Modify sofsip cli to work with PCMA instead of PCMA
2009/9/14 fabien comte comte_fab...@yahoo.fr I want sofsip cli works with PCMA instead of PCMU. How to modify it ? Just in case you have not figured out it yet: you have to change the RTP payload type on m= line to 8, too. The rtpmap should work, provided the payload number on m=audio line and a=rtpmap are same, at least in principle, but most implementations just look the payload type number for well-known codecs like PCMU or PCMA. I modified ssc_media_gst.c [...] I tryed - self-sm_pt = 8; /* PT=0 = PCMA */ /* step: initialize the PT-caps hash table */ pt_caps = gst_caps_new_simple (application/x-rtp, clock-rate, G_TYPE_INT, 8000, encoding-name, G_TYPE_STRING, PCMA, NULL); /* step: describe capabilities in SDP terms */ /* support only G711/PCMA */ caps_sdp_str = su_strcat(home, caps_sdp_str, v=0\r\n m=audio 0 RTP/AVP 0\r\n a=rtpmap:8 PCMA/8000\r\n); m=audio 0 RTP/AVP 8\r\n a=rtpmap:8 PCMA/8000\r\n); *dest = strdup(caps_sdp_str); su_free(home, caps_sdp_str); It does not work (FAILED with 501 Not Implemented Yet) -- Pekka.Pessi mail at nokia.com -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Wrong answer from sofia-sip
2009/8/26 Bernhard Suttner sutt...@comdasys.com: attached the network trace. You have to decode the first paket as SIP in Wireshark (destination ip). Contact is used when the callee send caller a BYE request (or OPTIONS or UPDATE or INFO etc...). The RFC 3261 (section 18.2.1) specifies how to send the response: unless you happen to have maddr parameter, the response is sent to the source address, which is 10.251.0.1. If you can include a maddr parameter in the incoming INVITE Sofia should send the response to the address specified in maddr parameter. So you could have request like this: INVITE sip:757-2...@10.251.0.101:12004 SIP/2.0 Via: SIP/2.0/UDP 10.27.0.60:5060;branch=z9hG4bK3745033304-3162;maddr=10.27.0.60 From: Aman DECT sip:7074...@10.23.0.203:5060;tag=shorUA_3745033458-3162 To: 757-2026 sip:757-2...@10.23.0.203 Contact: Aman DECT sip:7074...@10.27.0.60:5060 ... and Sofia would respond like this (to 10.27.0.60): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.27.0.60:5060;branch=z9hG4bK3745033304-3162;maddr=10.27.0.60;received=10.251.b0.1 From: Aman DECT sip:7074...@10.23.0.203:5060;tag=shorUA_3745033458-3162 To: 757-2026 sip:757-2...@10.23.0.203 ... --Pekka -Ursprüngliche Nachricht- Von: Michael Jerris [mailto:m...@jerris.com] Gesendet: Mittwoch, 26. August 2009 16:53 An: sofia-sip-devel Betreff: Re: [Sofia-sip-devel] Wrong answer from sofia-sip Can you paste the actual sip trace here. Mike On Aug 26, 2009, at 6:45 AM, Bernhard Suttner wrote: Hi, I have the following problem (using sofia sip 1.12.9): A (non sofia-sip) sends INVITE to B (user agent with sofia-sip) B send back a Trying B send Ringing and then a 200 OK Most important data for the INVITE from A: Src IP: 10.251.0.1 Dest IP: 10.251.0.101 Contact: 10.27.0.60 The Trying generated from sofia-sip has the data: Src IP: 10.251.0.101 Dest IP: 10.251.0.1 VIA: SIP/2.0/UDP 10.27.0.60;branch=ydfasfasfsf;received=10.251.0.1 The problem is, that B use the IP 10.251.0.1 as destination and not the contact of the INVITE (that would be 10.27.0.60). Also in the RINGING and 200 OK sofia-sip will use the Src-IP and not the Contact addr. Can I somehow configure sofia-sip in that way, that it will use the CONTACT addr instead of the Src IP? Thanks in advance! Best regards, Bernhard Suttner -- Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel -- Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel -- Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel -- Pekka.Pessi mail at nokia.com -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] Mac OS X 10.6 build problems
Any news on this? Perhaps you have something in /usr/local/lib linking to iconv? What is the intl library? Try grep -l iconv /usr/lib/* /usr/local/lib/* ... --Pekka 2009/9/1 Michael Jerris m...@jerris.com: I am chasing down this same issue on my box. I should have fixes sometime this week and will push patches for them when I have them. Note, I did have a successfull snow lep build a few weeks back before they did gold master on a different box, still trying to chase down what changed sense then. Mike On Aug 31, 2009, at 7:03 PM, Daniel Corbe wrote: gmake[3]: Leaving directory `/usr/local/src/sofia-sip-1.12.10/libsofia-sip-ua-glib/su-glib' gmake[3]: Entering directory `/usr/local/src/sofia-sip-1.12.10/libsofia-sip-ua-glib' /bin/sh ../libtool --tag=CC --mode=link gcc -g -O2 -o libsofia-sip-ua-glib.la -rpath /usr/local/lib -version-info 3:0:0 su-glib/libsu-glib.la -L/usr/local/lib -lglib-2.0 -lintl -lssl -lcrypto -lz -lpthread mkdir .libs grep: /usr/lib/libiconv.la: No such file or directory sed: /usr/lib/libiconv.la: No such file or directory libtool: link: `/usr/lib/libiconv.la' is not a valid libtool archive gmake[3]: *** [libsofia-sip-ua-glib.la] Error 1 gmake[3]: Leaving directory `/usr/local/src/sofia-sip-1.12.10/libsofia-sip-ua-glib' gmake[2]: *** [all-recursive] Error 1 gmake[2]: Leaving directory `/usr/local/src/sofia-sip-1.12.10/libsofia-sip-ua-glib' gmake[1]: *** [all-recursive] Error 1 gmake[1]: Leaving directory `/usr/local/src/sofia-sip-1.12.10' gmake: *** [all] Error 2 sh-3.2# cd /usr/lib sh-3.2# ls *iconv* libiconv.2.4.0.dylib libiconv.2.dylib libiconv.dylib -- Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel -- Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel -- Pekka.Pessi mail at nokia.com -- Come build with us! The BlackBerryreg; Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9#45;12, 2009. Register now#33; http://p.sf.net/sfu/devconf ___ 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: call termination after INFO-timeout
2009/7/7 Stefan Eckel stefan.ec...@c4b.de: Some time ago I posted a question about handling the timeout problem with INFO messages. See: http://article.gmane.org/gmane.comp.telephony.sofia-sip.devel/3402 We now found a solution, with a simple modification: ... I don’t know whether the solution is suitable for public project. We havn’t found any disadvantates so far. Sorry about that, sip_response_terminates_dialog() and how it is used by nua should be documented better. INFO is somewhat problematic, the Freeswitch project have turned it as a generic.purpose method (iow it behaves pretty much like MESSAGE, and it does not anything to do with INVITE-initiated session. Probably what is best way to go in a public project. -- Pekka.Pessi mail at nokia.com -- Enter the BlackBerry Developer Challenge This is your chance to win up to $100,000 in prizes! For a limited time, vendors submitting new applications to BlackBerry App World(TM) will have the opportunity to enter the BlackBerry Developer Challenge. See full prize details at: http://p.sf.net/sfu/Challenge ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
