[Sofia-sip-devel] some more issues......
hi,I installed Fedora core 5 and restarted the sofia sip installation process again.after having installed various gstreamer packegs (the devel ones, needed by sofsip_cli) here is what it says to me:[EMAIL PROTECTED] sofsip-cli]# sofsip_cli sip:[EMAIL PROTECTED]** Message: This program is linked against GStreamer 0.10.3** (sofsip_cli:20958): DEBUG: priv_verify_required_elements:191 ** Message: Verifying GST element "alawenc" -> OK ** Message: Verifying GST element "alawdec" -> OK** Message: Verifying GST element "dynudpsink" -> OK** Message: Verifying GST element "udpsrc" -> OK** (sofsip_cli:20958): DEBUG: ssc_media_class_init:124 ** (sofsip_cli:20958): DEBUG: ssc_media_gst_class_init:138** (sofsip_cli:20958): DEBUG: ssc_media_init:167** Message: Selecting media implementation: gstreamer** (sofsip_cli:20958): DEBUG: priv_static_capabilities_gst sofsip> UA: unknown event 23: 200 OK ::tag_null: 0sofsip> UA: nua_r_getparams: 200 OK sip::from: sip:[EMAIL PROTECTED] sip::from_str: " sip:[EMAIL PROTECTED]" nua::retry_count: 3 nua::max_subscriptions: 20 nua::enableInvite: true nua::autoAlert: true nua::early_media: false nua::autoAnswer: false nua::autoACK: true nua::invite_timer: 120 nua::session_timer: 0 nua::min_se: 120 nua::session_refresher: 0 nua::update_refresh: false nua::enableMessage: true nua::enableMessenger: false nua::callee_caps: false nua::media_features: false nua::service_route_enable: true nua::path_enable: true nua::substate: 2 sip::supported: timer, 100rel sip::supported_str: "timer, 100rel" sip::allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE sip::allow_str: "INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE" sip::user_agent: sofia-sip/1.11.8 sip::user_agent_str: "sofia-sip/1.11.8" nua::outbound: "natify" nua::keepalive: 12 nua::media_enable: true nua::registrar: nta::contact: nta::udp_mtu: 1300 nta::sip_t1: 500 nta::sip_t2: 4000 nta::sip_t4: 5000 nta::sip_t1x64: 32000 nta::debug_drop_prob: 0 nta::default_proxy: nta::aliases: nta::sipflags: 2 soa::caps_sdp: v=0 o=- 7835482480909229261 8743008824963640866 IN IP4 163.162.45.195s=-c=IN IP4 myipt=0 0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 soa::caps_sdp_str: "v=0 o=- 7835482480909229261 8743008824963640866 IN IP4 163.162.45.195s=-c=IN IP4 myipt=0 0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000" soa::user_sdp: v=0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 soa::user_sdp_str: "v=0m=audio 0 RTP/AVP 0a=rtpmap:0 PCMU/8000" soa::local_sdp_str: soa::af: 3 soa::srtp_enable: false soa::srtp_confidentiality: false soa::srtp_integrity: false ::tag_null: 0sofsip> tport_stun_bind_cb: stun_errortport_stun_bind_cb: stun_errordoes anyone know what it means?I haven't understood why it searchs for a stun server since I didn't specified it via the --stun option. the other strange thing is that the media port is set to zero!!best regardsluca
Re: [Sofia-sip-devel] some more issues......
I see the same log, but which looks ok. After registration, it crashed if I originate a call, and it cannot accept the incoming call. I'm wonding if you see the same thing. On 5/26/06, Luca Colantonio <[EMAIL PROTECTED]> wrote: hi,I installed Fedora core 5 and restarted the sofia sip installation process again.after having installed various gstreamer packegs (the devel ones, needed by sofsip_cli) here is what it says to me: [EMAIL PROTECTED] sofsip-cli]# sofsip_cli sip:[EMAIL PROTECTED]** Message: This program is linked against GStreamer 0.10.3 ** (sofsip_cli:20958): DEBUG: priv_verify_required_elements:191 ** Message: Verifying GST element "alawenc" -> OK ** Message: Verifying GST element "alawdec" -> OK** Message: Verifying GST element "dynudpsink" -> OK ** Message: Verifying GST element "udpsrc" -> OK** (sofsip_cli:20958): DEBUG: ssc_media_class_init:124 ** (sofsip_cli:20958): DEBUG: ssc_media_gst_class_init:138** (sofsip_cli:20958): DEBUG: ssc_media_init:167 ** Message: Selecting media implementation: gstreamer** (sofsip_cli:20958): DEBUG: priv_static_capabilities_gst sofsip> UA: unknown event 23: 200 OK ::tag_null: 0sofsip> UA: nua_r_getparams: 200 OK sip::from: sip:[EMAIL PROTECTED] sip::from_str: " sip:[EMAIL PROTECTED]" nua::retry_count: 3 nua::max_subscriptions: 20 nua::enableInvite: true nua::autoAlert: true nua::early_media: false nua::autoAnswer: false nua::autoACK: true nua::invite_timer: 120 nua::session_timer: 0 nua::min_se: 120 nua::session_refresher: 0 nua::update_refresh: false nua::enableMessage: true nua::enableMessenger: false nua::callee_caps: false nua::media_features: false nua::service_route_enable: true nua::path_enable: true nua::substate: 2 sip::supported: timer, 100rel sip::supported_str: "timer, 100rel" sip::allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE sip::allow_str: "INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE" sip::user_agent: sofia-sip/1.11.8 sip::user_agent_str: "sofia-sip/1.11.8" nua::outbound: "natify" nua::keepalive: 12 nua::media_enable: true nua::registrar: nta::contact: nta::udp_mtu: 1300 nta::sip_t1: 500 nta::sip_t2: 4000 nta::sip_t4: 5000 nta::sip_t1x64: 32000 nta::debug_drop_prob: 0 nta::default_proxy: nta::aliases: nta::sipflags: 2 soa::caps_sdp: v=0 o=- 7835482480909229261 8743008824963640866 IN IP4 163.162.45.195s=-c=IN IP4 myipt=0 0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 soa::caps_sdp_str: "v=0o=- 7835482480909229261 8743008824963640866 IN IP4 163.162.45.195 s=-c=IN IP4 myipt=0 0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000" soa::user_sdp: v=0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000 soa::user_sdp_str: "v=0m=audio 0 RTP/AVP 0 a=rtpmap:0 PCMU/8000" soa::local_sdp_str: soa::af: 3 soa::srtp_enable: false soa::srtp_confidentiality: false soa::srtp_integrity: false ::tag_null: 0sofsip> tport_stun_bind_cb: stun_error tport_stun_bind_cb: stun_errordoes anyone know what it means?I haven't understood why it searchs for a stun server since I didn't specified it via the --stun option. the other strange thing is that the media port is set to zero!! best regards luca
Re: [Sofia-sip-devel] some more issues......
On Fri, 26 May 2006, Luca Colantonio wrote: I installed Fedora core 5 and restarted the sofia sip installation process again. after having installed various gstreamer packegs (the devel ones, needed by sofsip_cli) here is what it says to me: Ok, let's see... [EMAIL PROTECTED] sofsip-cli]# sofsip_cli sip:[EMAIL PROTECTED] [...] ** Message: This program is linked against GStreamer 0.10.3 Ok, this looks good. ** Message: Selecting media implementation: gstreamer The plain gstreamer media implementation is chosen. sofsip> tport_stun_bind_cb: stun_error tport_stun_bind_cb: stun_error [...] does anyone know what it means? I haven't understood why it searchs for a stun server since I didn't specified it via the --stun option. You can ignore these. Sofsip-cli will do a DNS SRV lookup to find a STUN server for your domain. In this case it didn't find one. You should now be able to make calls "i sip:[EMAIL PROTECTED]" the other strange thing is that the media port is set to zero!! best regards That's just the static capabilities (reported if someone sends you a SIP OPTIONS request). So port zero is correct here. -- under work: Sofia-SIP at http://sofia-sip.sf.net --- All the advantages of Linux Managed Hosting--Without the Cost and Risk! Fully trained technicians. The highest number of Red Hat certifications in the hosting industry. Fanatical Support. Click to learn more http://sel.as-us.falkag.net/sel?cmd=lnk&kid=107521&bid=248729&dat=121642 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] some more issues with SDP negotiation
Hello!
In order of experimenting with SOFIA-SIP i have found some more problems
with SDP negotiation:
1. If remote peer modifying a session, SDP answer does not always contain
appropriate with offer codec.
2. Using SOA_RTP_SELECT_SINGLE brings to removing "telephone-event"
from rtp payloads in SDP answer.
There tport log with cisco ATA-186 (outgoing call from SOFIA-SIP application
to ATA-186, then ATA-186 decide switch to upspeed mode in order to receive
fax):
send 825 bytes to udp/[192.168.138.60]:5060 at 07:41:06.244585:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.138.65;rport;branch=z9hG4bKcQ6rvQgmrHjDg
Max-Forwards: 70
From: ;tag=cZ83ZvjKe5a9a
To:
Call-ID: 553cfe7e-98b5-1200-c88d-85c13a5d7ed1
CSeq: 30766410 INVITE
Contact:
User-Agent: TAU-32.IP v1.1 with sofia-sip/1.12.1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 260
v=0
o=- 8063427506184445420 8596425879938756690 IN IP4 192.168.138.65
s=-
c=IN IP4 192.168.138.65
t=0 0
m=audio 23000 RTP/AVP 4 8 0 101
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
recv 730 bytes from udp/[192.168.138.60]:5060 at 07:41:15.605089:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.138.65;rport=5060;branch=z9hG4bKcQ6rvQgmrHjDg
Record-Route:
From: ;tag=cZ83ZvjKe5a9a
To: ;tag=895631890
Call-ID: 553cfe7e-98b5-1200-c88d-85c13a5d7ed1
CSeq: 30766410 INVITE
Contact:
Server: Cisco ATA 186 v3.1.0 atasip (040211A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Type: application/sdp
Content-Length: 197
v=0
o=4000 3260 3260 IN IP4 192.168.138.70
s=ATA186 Call
c=IN IP4 192.168.138.70
t=0 0
m=audio 16384 RTP/AVP 4 101
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
recv 829 bytes from udp/[192.168.138.60]:5060 at 07:41:17.539201:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.138.60:5060;rport;branch=z9hG4bKb64c330c7e5efbac.3
Via: SIP/2.0/UDP 192.168.138.70:5060
From: sip:[EMAIL PROTECTED];tag=895631890
To: sip:[EMAIL PROTECTED];tag=cZ83ZvjKe5a9a
Call-ID: 553cfe7e-98b5-1200-c88d-85c13a5d7ed1
CSeq: 1 INVITE
Contact:
User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A)
Expires: 10
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Record-Route:
Content-Type: application/sdp
Content-Length: 223
v=0
o=4000 3454 3454 IN IP4 192.168.138.70
s=ATA186 Call
c=IN IP4 192.168.138.70
t=0 0
m=audio 16384 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
send 779 bytes to udp/[192.168.138.60]:5060 at 07:41:17.547193:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.138.60:5060;rport=5060;branch=z9hG4bKb64c330c7e5efbac.3
Via: SIP/2.0/UDP 192.168.138.70:5060
Record-Route:
From: sip:[EMAIL PROTECTED];tag=895631890
To: sip:[EMAIL PROTECTED];tag=cZ83ZvjKe5a9a
Call-ID: 553cfe7e-98b5-1200-c88d-85c13a5d7ed1
CSeq: 1 INVITE
Contact:
User-Agent: TAU-32.IP v1.1 with sofia-sip/1.12.1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 156
v=0
o=- 8063427506184445420 8596425879938756691 IN IP4 192.168.138.65
s=-
c=IN IP4 192.168.138.65
t=0 0
m=audio 23000 RTP/AVP 4
a=rtpmap:4 G723/8000
...
To solve this, i have suggest some changes in soa_static.c:
*** soa_static.c2006-07-27 22:14:37.0 +0700
--- soa_static.c.new2006-09-05 18:26:27.286170880 +0700
***
*** 497,502
--- 497,503
{
sdp_media_t *sm;
sdp_media_t const *rm;
+ sdp_rtpmap_t *next;
for (sm = session->sdp_media, rm = remote->sdp_media;
sm && rm;
***
*** 515,522
if (common_codecs == 0)
;
else if (ss->ss_rtp_select == SOA_RTP_SELECT_SINGLE) {
! if (sm->m_rtpmaps)
! sm->m_rtpmaps->rm_next = NULL;
}
else if (ss->ss_rtp_select == SOA_RTP_SELECT_COMMON) {
soa_sdp_select_rtpmap(&sm->m_rtpmaps, rm->m_rtpmaps);
--- 516,528
if (common_codecs == 0)
;
else if (ss->ss_rtp_select == SOA_RTP_SELECT_SINGLE) {
! for (next=sm->m_rtpmaps; next; next=next->rm_next)
! if (!str0cmp("telephone-event", next->rm_encoding)) {
! next->rm_next = NULL;
! break;
! }
! if (sm->m_rtpmaps && sm->m_rtpmaps != next)
! sm->m_rtpmaps->rm_next = next;
}
else if (ss->ss_rtp_select == SOA_RTP_SELECT_COMMON) {
soa_sdp_select_rtpmap(&sm->m_rtpmaps, rm->m_rtpmaps);
***
*** 539,545
sdp_media_t *m, **m
Re: [Sofia-sip-devel] some more issues with SDP negotiation
Hi, On 9/5/06, Legostayev Denis <[EMAIL PROTECTED]> wrote: > In order of experimenting with SOFIA-SIP i have found some more problems > with SDP negotiation: > > 1. If remote peer modifying a session, SDP answer does not always contain > appropriate with offer codec. Uh-oh. Thanks for the patch. > 2. Using SOA_RTP_SELECT_SINGLE brings to removing "telephone-event" > from rtp payloads in SDP answer. I think there are also other codecs that could be selected along with single codec, like CN codec. Perhaps a list of codecs, specified by yet an another tag. Pekka -- Pekka.Pessi mail at nokia.com - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
Re: [Sofia-sip-devel] some more issues with SDP negotiation
On 9/5/06, Pekka Pessi <[EMAIL PROTECTED]> wrote:
On 9/5/06, Legostayev Denis <[EMAIL PROTECTED]> wrote:
> 2. Using SOA_RTP_SELECT_SINGLE brings to removing "telephone-event"
> from rtp payloads in SDP answer.
I think there are also other codecs that could be selected along with
single codec, like CN codec. Perhaps a list of codecs, specified by
yet an another tag.
I added SOATAG_AUDIO_AUX() for that purpose.
Denis, could you test the attached patch (against 1.12.1)?
--
Pekka.Pessi mail at nokia.com
soa-codec.darcs
Description: Binary data
Wed Sep 6 00:07:17 EEST 2006 [EMAIL PROTECTED]
* test_soa.c: testing codec negotiation when new codecs are added to previously rejected media.
Wed Sep 6 00:05:40 EEST 2006 [EMAIL PROTECTED]
* test_soa.c: testing SOATAG_AUDIO_AUX().
Wed Sep 6 00:04:54 EEST 2006 [EMAIL PROTECTED]
* soa_static.c: added new tag SOATAG_AUDIO_AUX() for listing auxiliary codecs.
Auxiliary codecs are now included in m=line even if there is no matching
codec at remote end. They are also ignored when selecting common codecs.
Wed Sep 6 00:03:10 EEST 2006 [EMAIL PROTECTED]
* soa_tag.h, soa_tag.c: added SOATAG_AUDIO_AUX().
Tue Sep 5 21:50:18 EEST 2006 [EMAIL PROTECTED]
* soa_static.c: always upgrade answer if offer has changed.
sdp.c: fixed sdp_media_cmp() - now include m_mode in comparison.
Based on bug report and initial patch submitted by Legostayev Denis.
--- old-sofia-sip/libsofia-sip-ua/soa/soa_static.c 2006-09-06 00:23:22.0 +0300
+++ new-sofia-sip/libsofia-sip-ua/soa/soa_static.c 2006-09-06 00:23:22.0 +0300
@@ -1,7 +1,7 @@
/*
* This file is part of the Sofia-SIP package
*
- * Copyright (C) 2005 Nokia Corporation.
+ * Copyright (C) 2006 Nokia Corporation.
*
* Contact: Pekka Pessi <[EMAIL PROTECTED]>
*
@@ -62,6 +62,8 @@
#include
#include
#include
+#include
+#include
#include "sofia-sip/soa.h"
#include
@@ -70,7 +72,6 @@
#define NONE ((void *)-1)
#define XXX assert(!"implemented")
-#define str0cmp(a, b) strcmp(a ? a : "", b ? b : "")
#if !HAVE_STRCASESTR
char *strcasestr(const char *haystack, const char *needle);
#endif
@@ -78,6 +79,7 @@
typedef struct soa_static_session
{
soa_session_t sss_session[1];
+ char *sss_audio_aux;
}
soa_static_session_t;
@@ -148,25 +150,66 @@
static int soa_static_set_params(soa_session_t *ss, tagi_t const *tags)
{
- return soa_base_set_params(ss, tags);
+ soa_static_session_t *sss = (soa_static_session_t *)ss;
+ char const *audio_aux = sss->sss_audio_aux;
+ int n, m;
+
+ n = tl_gets(tags,
+ SOATAG_AUDIO_AUX_REF(audio_aux),
+ TAG_END());
+ if (n < 0)
+return n;
+
+ if (str0casecmp(audio_aux, sss->sss_audio_aux)) {
+char *s = su_strdup(ss->ss_home, audio_aux), *tbf = sss->sss_audio_aux;
+if (s == NULL && audio_aux != NULL)
+ return -1;
+sss->sss_audio_aux = s;
+if (tbf)
+ su_free(ss->ss_home, tbf);
+ }
+
+ m = soa_base_set_params(ss, tags);
+ if (m < 0)
+return m;
+
+ return n + m;
}
static int soa_static_get_params(soa_session_t const *ss, tagi_t *tags)
{
-
- return soa_base_get_params(ss, tags);
+ soa_static_session_t *sss = (soa_static_session_t *)ss;
+
+ int n, m;
+
+ n = tl_tgets(tags,
+ SOATAG_AUDIO_AUX(sss->sss_audio_aux),
+ TAG_END());
+ if (n < 0)
+return n;
+
+ m = soa_base_get_params(ss, tags);
+ if (m < 0)
+return m;
+
+ return n + m;
}
static tagi_t *soa_static_get_paramlist(soa_session_t const *ss,
tag_type_t tag, tag_value_t value,
...)
{
+ soa_static_session_t *sss = (soa_static_session_t *)ss;
+
ta_list ta;
tagi_t *tl;
ta_start(ta, tag, value);
- tl = soa_base_get_paramlist(ss, TAG_NEXT(ta_args(ta)));
+ tl = soa_base_get_paramlist(ss,
+ TAG_IF(sss->sss_audio_aux,
+ SOATAG_AUDIO_AUX(sss->sss_audio_aux)),
+ TAG_NEXT(ta_args(ta)));
ta_end(ta);
@@ -273,6 +316,37 @@
return rm != NULL;
}
+/** Check if codec is in auxiliary list */
+int soa_sdp_is_auxiliary_codec(sdp_rtpmap_t const *rm, char const *auxiliary)
+{
+ char const *codec;
+ size_t clen, alen;
+ char const *match;
+
+ if (!rm || !rm->rm_encoding || !auxiliary)
+return 0;
+
+ codec = rm->rm_encoding;
+
+ clen = strlen(codec), alen = strlen(auxiliary);
+
+ if (clen > alen)
+return 0;
+
+ for (match = auxiliary;
+ (match = strcasestr(match, codec));
+ match = match + 1) {
+if (IS_ALPHANUM(match[clen]) || match[clen] == '-')
+ continue;
+if (match != auxiliary &&
+ (IS_ALPHANUM(match[-1]) || match[-1] == '-'))
+ continue;
+return 1;
+ }
+
+ return 0;
+}
+
/** Find first matching media in table. */
sdp_media_t *soa_sdp_matching(soa_session_t *ss,
@@ -283,6 +357,14 @@
int i, j = -1;
sdp_media_t *m;
sdp_rtpmap_t const *rm;
+ soa_static_session_t *sss = (soa_static_session_t *)ss;
+ char const *auxiliary;
+
+ auxiliary = with->m_type == sdp_media_audio ? sss->sss_audio_aux
Re: [Sofia-sip-devel] some more issues with SDP negotiation
>> I think there are also other codecs that could be selected along with
>> single codec, like CN codec. Perhaps a list of codecs, specified by
>> yet an another tag.
>
> I added SOATAG_AUDIO_AUX() for that purpose.
>
> Denis, could you test the attached patch (against 1.12.1)?
What about auxiliary audio - it seems to work fine, but there still problems
with respond to re-INVITE:
send 825 bytes to udp/[192.168.138.60]:5060 at 04:49:40.885611:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.138.65;rport;branch=z9hG4bK7F4QU3rUmXmUK
Max-Forwards: 70
From: ;tag=7a93c1ZSmerrg
To:
Call-ID: 62afc021-989d-1200-f481-8d6be3d4a18d
CSeq: 30761267 INVITE
Contact:
User-Agent: TAU-32.IP v1.1 with sofia-sip/1.12.1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 260
v=0
o=- 4055417593070913274 8147886360907374363 IN IP4 192.168.138.65
s=-
c=IN IP4 192.168.138.65
t=0 0
m=audio 23000 RTP/AVP 4 8 0 101
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
recv 732 bytes from udp/[192.168.138.60]:5060 at 04:49:50.232295:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.138.65;rport=5060;branch=z9hG4bK7F4QU3rUmXmUK
Record-Route:
From: ;tag=7a93c1ZSmerrg
To: ;tag=202699017
Call-ID: 62afc021-989d-1200-f481-8d6be3d4a18d
CSeq: 30761267 INVITE
Contact:
Server: Cisco ATA 186 v3.1.0 atasip (040211A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Type: application/sdp
Content-Length: 199
v=0
o=4000 15814 15814 IN IP4 192.168.138.70
s=ATA186 Call
c=IN IP4 192.168.138.70
t=0 0
m=audio 16384 RTP/AVP 4 101
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
recv 831 bytes from udp/[192.168.138.60]:5060 at 04:49:52.185998:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.138.60:5060;rport;branch=z9hG4bK38aa6edee6d55bac.3
Via: SIP/2.0/UDP 192.168.138.70:5060
From: sip:[EMAIL PROTECTED];tag=202699017
To: sip:[EMAIL PROTECTED];tag=7a93c1ZSmerrg
Call-ID: 62afc021-989d-1200-f481-8d6be3d4a18d
CSeq: 1 INVITE
Contact:
User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A)
Expires: 10
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Record-Route:
Content-Type: application/sdp
Content-Length: 225
v=0
o=4000 16010 16010 IN IP4 192.168.138.70
s=ATA186 Call
c=IN IP4 192.168.138.70
t=0 0
m=audio 16384 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
send 811 bytes to udp/[192.168.138.60]:5060 at 04:49:52.194398:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.138.60:5060;rport=5060;branch=z9hG4bK38aa6edee6d55bac.3
Via: SIP/2.0/UDP 192.168.138.70:5060
Record-Route:
From: sip:[EMAIL PROTECTED];tag=202699017
To: sip:[EMAIL PROTECTED];tag=7a93c1ZSmerrg
Call-ID: 62afc021-989d-1200-f481-8d6be3d4a18d
CSeq: 1 INVITE
Contact:
User-Agent: TAU-32.IP v1.1 with sofia-sip/1.12.1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 188
v=0
o=- 4055417593070913274 8147886360907374365 IN IP4 192.168.138.65
s=-
c=IN IP4 192.168.138.65
t=0 0
m=audio 23000 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
...
The majority problem is that the answer on offer in second round is based on
the last negotiated SDP, but not user supplied
(soa_base_set_params does not increment ss_user_version number in most
cases, which is compared in offer_answer_step).
Unfortunately i have not good idea how to solve this.
In my previous patch soa_sdp_upgrade was called in case if remote version
have been changed:
/* Step B: upgrade local SDP (add m= lines to it) */
case generate_answer:
/* Upgrade local SDP based on remote SDP */
if (ss->ss_local_user_version == user_version &&
ss->ss_local_remote_version == remote_version)
break;
if (ss->ss_local_user_version != user_version ||
+ss->ss_local_remote_version != remote_version ||
soa_sdp_upgrade_is_needed(local, remote)) {
if (local != local0)
*local0 = *local, local = local0;
SU_DEBUG_7(("soa_static(%p, %s): %s\n", ss, by,
"upgrade with remote description"));
soa_sdp_upgrade(ss, tmphome, local, user, remote);
}
...
but that is of course is not good decision.
May be there should be more precision test to increment ss_user_version or
may be it always should be incremented when user pass SDP to SOA?
In last case i think there should not be a problems: if the user want to use
last negotiated sessi
Re: [Sofia-sip-devel] some more issues with SDP negotiation
On 9/6/06, Legostayev Denis <[EMAIL PROTECTED]> wrote:
What about auxiliary audio - it seems to work fine, but there still problems
with respond to re-INVITE:
Hmph. I'm afraid your original patch is the best approach.
I was afraid it would mess with media line matching. Let's see if
anyone complains. ;)
Here is a patch against previous one, the changes are also committed
into darcs repo at http://sofia-sip.org/repos/sofia-sip/
--
Pekka.Pessi mail at nokia.com
Wed Sep 6 17:10:17 EEST 2006 [EMAIL PROTECTED]
* soa_static.c: fixed problem of a new offer not including active codec.
test_soa.c: added test for dropping codecs from new offer.
Based on patch by Legostayev Denis.
diff -rN -u -udp old-sofia-sip/libsofia-sip-ua/soa/soa_static.c new-sofia-sip/libsofia-sip-ua/soa/soa_static.c
--- old-sofia-sip/libsofia-sip-ua/soa/soa_static.c 2006-09-06 17:11:20.0 +0300
+++ new-sofia-sip/libsofia-sip-ua/soa/soa_static.c 2006-09-06 17:11:20.0 +0300
@@ -999,8 +999,7 @@ static int offer_answer_step(soa_session
if (ss->ss_local_user_version == user_version &&
ss->ss_local_remote_version == remote_version)
break;
-if (ss->ss_local_user_version != user_version ||
- soa_sdp_upgrade_is_needed(local, remote)) {
+if (1) {
if (local != local0)
*local0 = *local, local = local0;
SU_DEBUG_7(("soa_static(%p, %s): %s\n", ss, by,
@@ -1062,10 +1061,9 @@ static int offer_answer_step(soa_session
break;
}
- /* Step E: Upgrade codecs */
+ /* Step E: Upgrade codecs by answer. */
switch (action) {
case process_answer:
- case generate_answer:
/* Upgrade local SDP based on remote SDP */
if (ss->ss_local_remote_version == remote_version)
break;
@@ -1080,6 +1078,7 @@ static int offer_answer_step(soa_session
}
break;
case generate_offer:
+ case generate_answer:
default:
break;
}
diff -rN -u -udp old-sofia-sip/libsofia-sip-ua/soa/test_soa.c new-sofia-sip/libsofia-sip-ua/soa/test_soa.c
--- old-sofia-sip/libsofia-sip-ua/soa/test_soa.c 2006-09-06 17:11:20.0 +0300
+++ new-sofia-sip/libsofia-sip-ua/soa/test_soa.c 2006-09-06 17:11:20.0 +0300
@@ -1015,6 +1015,90 @@ int test_codec_selection(struct context
TEST_1(m = m->m_next); TEST_1(!m->m_rejected);
TEST_1(!m->m_next);
+ /* -- */
+ /* A drops GSM support */
+
+ TEST_1(soa_set_params(a,
+ SOATAG_USER_SDP_STR(
+"v=0\r\n"
+"o=left 219498671 2 IN IP4 127.0.0.2\r\n"
+"c=IN IP4 127.0.0.2\r\n"
+"m=audio 5008 RTP/AVP 0 8 96 127\r\n"
+"a=rtpmap:96 G729/8000\n"
+"a=rtpmap:127 CN/8000\n"
+"m=video 5010 RTP/AVP 31 34\r\n"
+"m=audio 6008 RTP/SAVP 3\n"
+),
+ TAG_END()));
+
+ /* B adds GSM to SRTP */
+ TEST_1(soa_set_params(b,
+ SOATAG_USER_SDP_STR(
+"v=0\r\n"
+"o=left 219498671 2 IN IP4 127.0.0.2\r\n"
+"c=IN IP4 127.0.0.2\r\n"
+"m=audio 5004 RTP/AVP 96 3 97 111\r\n"
+"a=rtpmap:96 G7231/8000\n"
+"a=rtpmap:97 G729/8000\n"
+"a=rtpmap:111 telephone-event/8000\n"
+"a=fmtp:111 0-15\n"
+"m=audio 6004 RTP/SAVP 96 3\n"
+"a=rtpmap:96 G729/8000\n"
+"m=video 5006 RTP/AVP 34\n"
+),
+ TAG_END()));
+
+ n = soa_generate_offer(a, 1, test_completed); TEST(n, 0);
+ n = soa_get_local_sdp(a, &a_sdp, &offer, &offerlen); TEST(n, 1);
+ TEST_1(offer != NULL && offer != NONE);
+ TEST_1(m = a_sdp->sdp_media); TEST_1(!m->m_rejected);
+ TEST_1(m = m->m_next); TEST_1(!m->m_rejected);
+ TEST_1(m = m->m_next); TEST_1(!m->m_rejected);
+ TEST_1(!m->m_next);
+ n = soa_set_remote_sdp(b, 0, offer, offerlen); TEST(n, 1);
+ n = soa_generate_answer(b, test_completed); TEST(n, 0);
+ /* Answer from B now accepts video */
+ n = soa_get_local_sdp(b, &b_sdp, &answer, &answerlen); TEST(n, 1);
+ TEST_1(answer != NULL && answer != NONE);
+ n = soa_set_remote_sdp(a, 0, answer, -1); TEST(n, 1);
+ n = soa_process_answer(a, test_completed); TEST(n, 0);
+ n = soa_get_local_sdp(a, &a_sdp, &offer, &offerlen); TEST(n, 1);
+
+ TEST_1(soa_is_complete(b));
+ TEST(soa_activate(b, NULL), 0);
+
+ TEST_1(soa_is_complete(a));
+ TEST(soa_activate(a, NULL), 0);
+
+ TEST(soa_is_audio_active(a), SOA_ACTIVE_SENDRECV);
+ TEST(soa_is_remote_audio_active(a), SOA_ACTIVE_SENDRECV);
+ TEST(soa_is_video_active(a), SOA_ACTIVE_SENDRECV);
+ TEST(soa_is_remote_video_active(a), SOA_ACTIVE_SENDRECV);
+
+ TEST_1(m = a_sdp->sdp_media); TEST_1(!m->m_rejected);
+ TEST_1(rm = m->m_rtpmaps); TEST(rm->rm_pt, 96);
+ TEST_S(rm->rm_encoding, "G729");
+ TEST_1(rm = rm->rm_next); TEST(rm->rm_pt, 127);
+ TEST_S(rm->rm_encoding, "CN");
+ TEST_1(m = m->m_next); TEST_1(!m->m_rejected);
+ TEST_1(rm = m->m_rtpmaps); TEST(rm->rm_pt, 34);
+ TEST_S(rm->rm_encoding, "H263");
+ TEST_1(m = m->m_next); TEST_1(!m->m_rejected);
+ TEST_1(!m->m_next);
+
+ TEST_1(m = b_sdp->sdp_media); TEST_1(!m->m_rejected);
+ TEST_1(rm = m->m_rtpmaps); TEST(rm->rm_pt, 96);
+ TEST_S(rm->rm
